- sr-users - kamailio.org

[Kamailio-Users] What happens with filled htables?
by catalina oancea 26 Aug '09

26 Aug '09

Hello I use the htable module a lot but the only problem, when I add a new entry in a htable, is when I will delete it. My question is: if the hash table is completely filled and I try to add a new value to it, do I get an error or is an old value automatically deleted to be able to write my new value? If an old value was automatically deleted whenever a new value is added, I wouldn't have to bother deleting the values I no longer need. Thanks Catalina

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[Serusers] translate message from TLS to UDP
by qiulei212 26 Aug '09

26 Aug '09

hi everyone: sorry to trouble you again! I use the SER as a proxy between a SIP-softphone and the asterisk. The message between the sip-phone and SER is TLS and i want to translant the message by UDP to my asterisk.HOW would i do? I have tried the forward_udp , t_relay_to_tls ,t_relay_to_udp and no useless. which function should i prefer? thanks!! qiulei212 2009-08-26

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[Serusers] problems with stun using uas
by lorenzo 25 Aug '09

25 Aug '09

hi! i'm the developer of a sip app that happens to be connecting to a sip server that's using SER. i'm having troublewith the way SER handles incoming calls to my stun-enabled ua (which is placed behind a router) I have two clients running on the same machine. one is my app, the other is Gizmo5. when i make an outgoing call from my app to the user i'm logged in with in Gizmo5, everything works. but when i call my app from the Gizmo5 app, sniffing the network traffic with wireshark, the only thing i see happening is the sip server sending a 100 Giving a try to the calling ua. The reason i'm posting this here, is that i know it's not a bug with my app, since everything works fine when i don't use stun. (note that i don't have direct access to ser's configuration) this is my app's REGISTER request (note that it has a *public ip*): REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 93.148.135.54:5060;rport;branch=z9hG4bK70207 Max-Forwards: 70 To: <sip:asymmetric@proxy01.sipphone.com> From: <sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK57274073 Call-ID: 557402852938(a)93.148.135.54 CSeq: 2 REGISTER Contact: <sip:asymmetric@93.148.135.54:5060;transport=udp> Expires: 3600 User-Agent: mjsip stack 1.6 Authorization: Digest username="asymmetric", realm="proxy01.sipphone.com", nonce="xxx", uri="sip:proxy01.sipphone.com", algorithm=md5, response="yyy" Content-Length: 0 and the response: SIP/2.0 200 OK Via: SIP/2.0/UDP 93.148.135.54:5060;rport=12810;branch=z9hG4bK70207 To: <sip:asymmetric@proxy01.sipphone.com>;tag=92390300a369f0d75803e369c733575e.50aa From: <sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK57274073 Call-ID: 557402852938(a)93.148.135.54 CSeq: 2 REGISTER Contact: <sip:asymmetric@93.148.135.54:5060;transport=udp>;expires=3600 Content-Length: 0 Gizmo5's app REGISTER request: REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-bbc1e2712d417352-1--d87543-;rport Max-Forwards: 70 Contact: <sip:17474492586@93.148.135.54:10251> To: <sip:17474492586@proxy01.sipphone.com> From: <sip:17474492586@proxy01.sipphone.com>;tag=f43be43c Call-ID: b1b4a060512d4b0c1250955005@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 2 REGISTER Expires: 1800 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER User-Agent: MacGizmo/2.0.02 (Gizmo-s2n1) Authorization: Digest username="17474492586",realm="proxy01.sipphone.com",nonce="xxx",uri="sip:proxy01.sipphone.com",response="yyy",algorithm=MD5 Content-Length: 0 and response: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-bbc1e2712d417352-1--d87543-;rport=10251;received=93.148.135.54 To: <sip:17474492586@proxy01.sipphone.com>;tag=92390300a369f0d75803e369c733575e.c70f From: <sip:17474492586@proxy01.sipphone.com>;tag=f43be43c Call-ID: b1b4a060512d4b0c1250955005@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 2 REGISTER P-Behind-NAT: Yes Contact: <sip:17474492586@93.148.135.54:10251>;expires=1800 Content-Length: 0 in the second case, the client gets the P-Behind-NAT flag set to Yes, and for a good reason. But then why is this all i get when trying to call the first ua? : SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-7fb07a028db5c137-1--d87543-;rport=10251;received=93.148.135.54 To: <sip:asymmetric@proxy01.sipphone.com> From: <sip:17474492586@proxy01.sipphone.com>;tag=bc135372 Call-ID: 88d41c746d51146e1250955219@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 1 INVITE P-Behind-NAT: Yes Content-Length: 0 whereas the inverse works: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKf3b1.bc014575.0 Via: SIP/2.0/UDP 93.148.135.54:5060;rport=13171;branch=z9hG4bK82654 Record-Route: <sip:198.65.166.131;lr;ftag=z9hG4bK90319798> Contact: <sip:17474492586@proxy01.sipphone.com> To: <sip:17474492586@proxy01.sipphone.com>;tag=6ae0e759 From: "asymmetric"<sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK90319798 Call-ID: 593542951976(a)93.148.135.54 CSeq: 1 INVITE User-Agent: MacGizmo/2.0.02 (Gizmo-s2n1) Content-Length: 0 CQBM: 244 -- thanks a lot! asymmetric

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Presence over NAT
by Andrew O. Zhukov 25 Aug '09

25 Aug '09

Is there any samples let's forwarded Subscribe fix the Contact: I mean Proxy URI not the customer mapped. Let's the followed NOTIFY go over proxy not P2P. Thanks in avance. Andrew O. Zhukov

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[Kamailio-Users] kamailio 3.0 - the time before freezing
by Daniel-Constantin Mierla 25 Aug '09

25 Aug '09

Hello everybody, during the last IRC devel meeting, we pinned end of summer to be the time to enter testing phase for the new major release - codenamed kamailio 3.0. There was quite a lot of brand new features, considering that most of devel effort was directed to integration. A try to collect new items is available at: http://sip-router.org/wiki/features/new-in-devel Of course, all ser features are available as well, resulting a large amount of new stuff in kamailio 3.0 vs kamailio 1.5.x. See: http://sip-router.org/benefits/ http://www.iptel.org/ser/features I propose to start the testing phase sometime next week. Meanwhile, let's see if something important was forgotten from kamailio core or tm. To my knowledge: - path support in tm - Andrei has it on his todo afaik - in a way or another, it must get in next release - drop compatibility - drop reply in reply_route - internally drop and exit are now differentiated by core, needs review and update of handling after running the routes - on my to-do Apart seas, all Kamailio modules are fully ugraded to new core - I started but lack of test env + busy summer resulted in delays - can be done during testing period, being just api updates. If you have discovered something else, let us know. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] exchanging CDR data between Asterisk servers
by Klaus Darilion 24 Aug '09

24 Aug '09

Hi! I have the following setup: PSTN<------>Asterisk<-----SIP-->Asterisk GW/LCR \ \ ... \ \ ... \ --SIP-->Asterisk \ ... ------->Asterisk The GW-Asterisk just does the gatewaying stuff and writes the CDRs for the billing system. The other Asterisk servers handle all the services (IVR, REGISTER, ...) Some scenarios require to write some additional data to the CDRs. For outgoing calls this is not a problem (I signal the extra data in a SIP header and set a CDR() variable in the GW asterisk). My problem are incoming calls. The "extra data" is only known to the service Asterisk, but the CDR is written by the GW Asterisk. Does anybody know a method how to signal the "extra data" during the call from the server Asterisk back to the GW Asterisk und put it into a CDR() variable? Regards Klaus

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[Kamailio-Users] TLS TCP error
by shehzad p 24 Aug '09

24 Aug '09

Hello, My Kamailio (tls version) is configured to handle TCP as well as TLS traffic. Making calls from softphone using TCP only works fine. But my TLS traffic has been stucked. I can see lots of error as below in messages file: ====================================================================== Aug 24 06:05:01 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Aug 24 06:05:01 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcp_send: connect failed Aug 24 06:05:01 ns110 /usr/local/sbin/kamailio[24273]: ERROR:tm:msg_send: tcp_send failed Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (110) Connection timed out Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcp_send: connect failed Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:tm:msg_send: tcp_send failed Aug 24 06:08:25 ns110 /usr/local/sbin/kamailio[24279]: ERROR:core:_tls_read: something wrong in SSL: 5 Aug 24 06:08:25 ns110 /usr/local/sbin/kamailio[24279]: ERROR:core:tcp_read_req: failed to read Aug 24 06:09:00 ns110 /usr/local/sbin/kamailio[24277]: INFO:core:tls_accept: client did not present a certificate ================================================================================= I am very confused, because with the same configuration, another kamailio server works very well without any error. I am not able to know what could be the issue in this server with same configuration. What could be the cause and resolution of this errors? Any help will be appreciated Thanks in advance, Mohammed Shehzad -- View this message in context: http://www.nabble.com/TLS-TCP-error-tp25114961p25114961.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] new PV class: $sel(name)
by Daniel-Constantin Mierla 24 Aug '09

24 Aug '09

Hello, there is a new PV class that provides access to SER select variables. A list with available selects can be found at: http://sip-router.org/wiki/cookbooks/selects/devel Therefore, all Kamailio modules relying on PV framework can get access to select values without any change. See example of usage from config at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#selects Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] sip phone behind the nat
by aiphie 24 Aug '09

24 Aug '09

Hi,body. I'm using the kamailio as a sip server,If in the local network ,It works very well ,But when i set the sip server in the internet, When starting voice chat,Two sip phone will receive the behind net ip address(sip phone's local ip address). How i can configure sip server(kamailio) let sip server send sip phone's peer ip address using front nat ip address(internet ip address)? Any body can give me how to realize it? Thank you! --Aiphie 2009-08-21 “欢乐家庭总动员”--第七届全球通演出季网络互动有惊喜，千元大奖送给您！移动新推GPRS流量优惠系列新套餐，手机上网省钱又省心！

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Re: [Kamailio-Users] sip phone behind the nat
by Daniel-Constantin Mierla 24 Aug '09

24 Aug '09

On 24.08.2009 9:38 Uhr, aiphie wrote: > hi,all > you probably forgot to cc to the mailing list. please do so all the time - there might be people that can help faster and better. > > I have use sed command to clear the comment which in front of regarding nat,But it's always use local network IP to fill session description protocol->connection information->connection address,I know sip server can rewrite this message item,Can you tell me how to let sip server rewrite it? > annex is my config file,please look what wrong in this? > It is very unlikely that one will debug a whole config file for you. You just need to investigate yourself and come to mailing list with specific questions. The media IP is updated with what RTPPproxy is listening on. What is the command used to start rtpproxy? Cheers, Daniel > thank you! > > --aiphie > > > > ----- Original Message ----- > From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> > To: "aiphie" <chengwu8(a)139.com> > Cc: <users(a)lists.kamailio.org> > Sent: Friday, August 21, 2009 6:55 PM > Subject: Re: [Kamailio-Users] sip phone behind the nat > > > >> Hello, >> >> On 21.08.2009 12:52 Uhr, aiphie wrote: >> >>> Hi,body. >>> I'm using the kamailio as a sip server,If in the local network ,It >>> works very well ,But when i set the sip server in the internet, When >>> starting voice chat,Two sip phone will receive the behind net ip >>> address(sip phone's local ip address). >>> How i can configure sip server(kamailio) let sip server send sip >>> phone's peer ip address using front nat ip address(internet ip address)? >>> Any body can give me how to realize it? >>> >> you need to enable nathelper module + appropriate lines in config file >> and install rtpptoxy. >> >> If you check the default config file you will see at the top some >> comments regarding nat traversal and what sed command you can use to >> enable it. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> * http://www.asipto.com/ >> > > -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Serusers] TLS module load error
by qiulei212 24 Aug '09

24 Aug '09

hello friends: I crashed when i compile the SER source and use the tls.so! I have install the openssl 0.9.8 in the default folder :/usr/local and install the ser-2.0.0-rc1 in the default too. run the command below: make group_include="standard" include_modules="tls" all make group_include="standard" include_modules="tls" installthe tls.so in the /usr/local/lib/ ser/modules folderthe I add the loadmodule "/usr/local/lib/ser/modules/tls.so" in the ser.cfg and start /usr/local/sbin/ serbut the ser start failed!I run the ser -c and found:ERROR: load_module: could not open module </usr/local/lib/ser/modules/tls.so>: /usr/local/lib/ser/modules/tls.so: undefined symbol: deflate I don't know Why and ask help for you ! Thanks a lot ! qiulei212 2009-08-21

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[Kamailio-Users] Choosing IP address when relaying request
by mayamatakeshi 20 Aug '09

20 Aug '09

Hello, I have 2 ip addresses being listened in my cfg file. Is it possible to set which ip address will be used when relaying a request? regards, takeshi

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[Kamailio-Users] Problem with BYE when using TCP or TLS between two Kamilio proxies
by Dubravko Caric 20 Aug '09

20 Aug '09

Hi all, I have a problem with routing BYE messages (when called UAC sends BYE) when connection between two Kamailio (1.5.1 with identical configurations) proxies is done by TLS/TCP. WORKING: UAC#1 <-- UDP/TCP/TLS --> Proxy#1 <--UDP--> Proxy#2 <-- UDP/TCP/TLS --> UAC#2 NOT WORKING: UAC#1 <-- UDP/TCP/TLS --> Proxy#1 <-- TCP or TLS --> Proxy#2 <-- UDP/TCP/TLS --> UAC#2 By looking at ngrep log I found out that when “UAC#1” sends INVITE, “Proxy#1” sends right “Contact” header (Contact:<sip:bob@IP_ADDRESS_OF_UAC#1:6126;transport=udp>) to “Proxy#2” , but “Proxy#2” sends wrong “Contact” header (Contact:<sip:bob@IP_ADDRESS_OF_PROXY#1:39879;transport=udp>) to “UAC#2”. When UAC#2 sends BYE, it is sends message with this wrong header and when it arrives on “Proxy#1” proxy doesn’t sends BYE message to UAC#1. Or when UACs with TCP, this are “Proxy#1” errors: DBG:core:tcp_send: no open tcp connection found, opening new one ERROR:core:tcp_blocking_connect: poll error: flags 18 ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused ERROR:core:tcpconn_connect: tcp_blocking_connect failed ERROR:core:tcp_send: connect failed ERROR:tm:msg_send: tcp_send failed ERROR:tm:t_forward_nonack: sending request failed DBG:tm:t_relay_to: t_forward_nonack returned error What bothers me is that when I call the other UAC, he can hangup the call (even when he is CALLED UAC), then I restart both proxies and it is other way around. I do not know if I described problem clearly but I am lost in where to look next so any help would be appreciated. Why proxy change Contact header when I switch from UDP to TCP between proxies? I'm not rewriting Contact header and only thing that I change in configuration when switching from UDP to TCP is this: if($rd=="example.com") { t_relay("tcp: example.com:5063"); exit; } Thanks a lot /dubravko

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[Kamailio-Users] limits per subscriber and per gateway
by Dmitri G. 20 Aug '09

20 Aug '09

Hello, First I would like to say thank you for the helpful members here, I have got useful tips and suggestions here. I am trying to implement some kind of "protection" for me, let me describe what is my problem at the moment. I know I ask a bit much questions here .) I would like to limit the number of incoming calls from my customers, but on a customer basis. I have a subscriber with an Asterisk, so it will be a bad idea to set a limit to 1 concurrent call for this customer, but limit 1 will work for a customer with an IP phone (remember the 100rel thing what I posted here? that was a workaround for this subscriber... I mean to remove 100rel from the supported header). So basically I would like to limit every subscriber, on a per customer basis. The next big thing is the lcr gateways, so where I send the calls. I have 2 peers, one of then have a limit for me, it's 20 concurrent calls. They don't limit me, so I can send out 30-40-50 calls to them, but they will charge me a big amount of Lei (Romanian currency) for every calls above 20 at the same time. They have freeswitch.... doesn't matter, I'm just saying they have freeswitch :) But they have very very good price(for 20 calls what i mentioned), so I would like to fill that "20 calls" space. I have asked them to give some limits for me, so refuse calls above 20 (so this way lcr can send out to the another gateway), but they don't want to do it (well, they need money I think) I would like to store values for these limits in mysql (column "limit" in subscriber table, and column "calllimit" in gw table, or I can store it in a separate table). Right now I store these values in a mysql table called "iplimits" I am not perfectly sure about how I can select these values from the table, and how can I "play" with these values. Even don't know how to retrieve the current number of concurrent calls per gateway or per subscriber. And what to do with these selected values? Store it in an avp and then do some checks against it? I am reading the module docs, especially sqlops, but I think I can't see the wood for the trees. I hope somebody can give me suggestions (or an example) about this. Kind regards, Dmitri

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[Kamailio-Users] inbound "failover"
by Dmitri G. 20 Aug '09

20 Aug '09

Sorry, I have accidentally sent this message to opensips mailing list, I am sending it now to the right place Hello, I would like to implement some kind of failover for my Asterisk, let me describe how I would like to see it. I registered 2 sip users with my Kamailio, 1020(a)domain.com and 1030(a)domain.com. I have added aliases to 1020 (aliases from 1021@domain com to 1029(a)domain.com). Right now calls to 1020-1029 goes well to 1020, it works fine. But I would like to do the following: If 1020 isn't registered with Kamailio (let's say if registration for 1020 is down in Kamailio, so AOR not found for 1020), it is possible to route calls to 1030? So route calls to 1030 only when registration for 1020 isn't active in Kamailio. I have tried manipulating with faillure_route, but without any luck. I hope it is clear what I would like to do :) Any help would be much appreciated. Kind regards, Dmitri

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Re: [Kamailio-Users] kamailio syslog logging
by Henning Westerholt 19 Aug '09

19 Aug '09

On Mittwoch, 19. August 2009, Jason Penton wrote: > Hey Henning, > > no, none of my log messages from the cfg file come into the log file. > However I do get the following: > > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 797031 local0.debug] > DBG:core:fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 596959 local0.debug] > DBG:core:fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16560 > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 362601 local0.debug] > DBG:core:fm_malloc_init: params (fbe00000, 33554432), start=fbe00000 > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 551434 local0.debug] > DBG:core:shm_mem_init_mallocs: success > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 123545 local0.info] > INFO:core:init_tcp: using /dev/poll as the TCP io watch method (auto > detected) > > weird that I get some messages but not the others. I have alot of LM_INFO > messages in my code that I dont see too????? Hi Jason, so it seems that you indeed get only DEBUG messages in the log file, so try the suggestion Alex gave to you - local0.* This should catch all. Cheers, Henning

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[Kamailio-Users] kamailio syslog logging
by Jason Penton 19 Aug '09

19 Aug '09

Hi All, I am having some problems logging Kamailio in debug mode using syslog. All is fine if I log to stderr. However, when I log to syslog I get a few dbug messages and then it stops -- I dont get any of my routing messages or anything???? i am suing solaris? Any suggestions Cheers Jason

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[Serusers] new_t: out of mem
by Roger Tan 19 Aug '09

19 Aug '09

hi. I tested my ser server which version is 0.9.7+cvs20080424 used sipps sending 1000 invite per sec. after about 10 sec, there was an out of mem error. Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: new_t: out of mem: Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: t_newtran: new_t failed Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: sip_msg_cloner: cannot allocate memory and my memlog is: Aug 18 10:11:02 ubuntu /sbin/ser[20787]: ----------------------------- Aug 18 10:11:02 ubuntu /sbin/ser[20787]: Memory status (shm): Aug 18 10:11:02 ubuntu /sbin/ser[20787]: qm_status (0xb5acf000): Aug 18 10:11:02 ubuntu /sbin/ser[20787]: heap size= 33554432 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: used= 1914592, used+overhead=2224736, free=31329696 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: max used (+overhead)= 33534720 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: dumping all alloc'ed. fragments: Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 0. N address=0xb5ad41a8 frag=0xb5ad41a0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 1. N address=0xb5ad41c8 frag=0xb5ad41c0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 2. N address=0xb5ad41e8 frag=0xb5ad41e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 3. N address=0xb5ad4208 frag=0xb5ad4200 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 4. N address=0xb5ad4228 frag=0xb5ad4220 size=4096 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 5. N address=0xb5ad5238 frag=0xb5ad5230 size=4096 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 6. N address=0xb5ad6248 frag=0xb5ad6240 size=2800 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 7. N address=0xb5ad6d48 frag=0xb5ad6d40 size=1680 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 8. N address=0xb5ad73e8 frag=0xb5ad73e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 9. N address=0xb5ad7408 frag=0xb5ad7400 size=1572864 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 10. N address=0xb5c57418 frag=0xb5c57410 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 11. N address=0xb5c57438 frag=0xb5c57430 size=320 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 12. N address=0xb5c57588 frag=0xb5c57580 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 13. N address=0xb5c575c8 frag=0xb5c575c0 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 14. N address=0xb5c57628 frag=0xb5c57620 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 15. N address=0xb5c57688 frag=0xb5c57680 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 16. N address=0xb5c576e8 frag=0xb5c576e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 17. N address=0xb5c57708 frag=0xb5c57700 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 18. N address=0xb5c57728 frag=0xb5c57720 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 19. N address=0xb5c57748 frag=0xb5c57740 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 20. N address=0xb5c57768 frag=0xb5c57760 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 21. N address=0xb5c57798 frag=0xb5c57790 size=131072 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 22. N address=0xb5c777a8 frag=0xb5c777a0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 23. N address=0xb5c777c8 frag=0xb5c777c0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 24. N address=0xb5c777e8 frag=0xb5c777e0 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 25. N address=0xb5c77818 frag=0xb5c77810 size=131072 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 60. N address=0xb5ca7948 frag=0xb5ca7940 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 61. N address=0xb5ca7988 frag=0xb5ca7980 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 62. N address=0xb5ca79a8 frag=0xb5ca79a0 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 63. N address=0xb5ca7a08 frag=0xb5ca7a00 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 64. N address=0xb5ca7a48 frag=0xb5ca7a40 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 65. N address=0xb5ca7a78 frag=0xb5ca7a70 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 66. N address=0xb5ca7a98 frag=0xb5ca7a90 size=65808 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: dumping free list stats : Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 1. fragments no.: 1232, unused: 0 ^I^I bucket size: 16 - 16 (first 16) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 38. fragments no.: 54, unused: 0 ^I^I bucket size: 608 - 608 (first 608) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 43. fragments no.: 54, unused: 0 ^I^I bucket size: 688 - 688 (first 688) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 44. fragments no.: 3991, unused: 0 ^I^I bucket size: 704 - 704 (first 704) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 57. fragments no.: 114, unused: 0 ^I^I bucket size: 912 - 912 (first 912) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 58. fragments no.: 4004, unused: 0 ^I^I bucket size: 928 - 928 (first 928) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 68. fragments no.: 46, unused: 0 ^I^I bucket size: 1088 - 1088 (first 1088) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 82. fragments no.: 109, unused: 0 ^I^I bucket size: 1312 - 1312 (first 1312) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 115. fragments no.: 34, unused: 0 ^I^I bucket size: 1840 - 1840 (first 1840) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 174. fragments no.: 3795, unused: 0 ^I^I bucket size: 2784 - 2784 (first 2784) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 186. fragments no.: 790, unused: 0 ^I^I bucket size: 2976 - 2976 (first 2976) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 187. fragments no.: 3820, unused: 0 ^I^I bucket size: 2992 - 2992 (first 2992) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 262. fragments no.: 1, unused: 0 ^I^I bucket size: 4192 - 4192 (first 4192) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 318. fragments no.: 1, unused: 0 ^I^I bucket size: 5088 - 5088 (first 5088) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: ----------------------------- I think it's not a memory leak but the so many small memory fragments, and those fragments were not combinated. How I solve this problem? Thanks. -- MSN: miao__tan(a)hotmail.com Email: m.roger.t(a)gmail.com

3 2

[Kamailio-Users] SIP Project for Beginner
by Sangmane 19 Aug '09

19 Aug '09

Hi all, I am new to SIP. I've downloaded Netbeans 6.7 and installed related plugins. When I tried to create an SIP Project, I couldn't find any SIP Categories in NB 6.7. Is there any basic guidance for SIP in Netbeans ? thank you very much. Sangmane -- View this message in context: http://www.nabble.com/SIP-Project-for-Beginner-tp25010377p25010377.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 2

[Kamailio-Users] remove 100rel from Supported header
by Dmitri G. 19 Aug '09

19 Aug '09

Hello, Is there a way to remove 100rel from Supported header (only this, and keep the others) by Kamailio? I am using Kamailio 1.5.1 notls. Kind regards, Dmitri

4 7

[Kamailio-Users] T38
by BERGANZ François 19 Aug '09

19 Aug '09

Hello, In a SIP Frame, how to see if it is a T38 frame? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 2

[Serusers] SER with a PSTN Gateway (pstn, cfg) vs called-party-BYE or Re-INVITE
by Frank Durda IV 19 Aug '09

19 Aug '09

I have noted that the pstn.cfg example (or anything built from its premise) doesn't work right for one very important situation, that hopefully someone has already discovered and has a good fix for. The documentation isn't helping. The situation: I have a call initiated by some SIP client, with the INVITE passing through SER and sent on to a PSTN gateway (the SIP to SS7/ISUP conversion occurs there). The call is in progress (post 200 OK) and both parties are chatting away. So far, so good. The problem: Trouble begins when the called-party (somewhere on the PSTN) hangs up first. What happens: This event triggers a SS7 REL message, and the SIP/SS7 switch providing the VOIP gateway service emits a BYE message, which gets sent to SER. SER receives the message and doesn't treat it as a reply (because it isn't), but instead it comes in the door at route {}. The pstn.cfg-like script quickly determines that this BYE message is coming from the gateway (correct) and skips all the authentication checks (good), and does a rewritehostport() using the PSTN gateway IP address (bad), and finally does a t_relay() (very bad). The BYE is now emitted by SER and sent back to the PSTN gateway switch, rather than on to wherever the INVITE originally came from, the device that needs to know about this BYE. This quickly degenerates into a small battle between PSTN gateway switch and SER, as the PSTN gateway isn't getting the answer back it is expecting in response to the BYE it sent to SER, and the PSTN gateway certainly doesn't want to see the message it just sent out come back over and over. Eventually various looping checks or timers kick in and the argument ends, but the calling system remains in the dark over the fact that the call is now over (other than getting a lot of silence). So that behavior is obviously wrong, but that's what the stock copy of pstn.cfg does here. SER obviously wants to be involved, because it added a Record-Route, but the script really isn't able to deal with the consequences. If the calling network was all SIP this would be merely annoying, but if it is another ISUP network (with a SIP leg in the middle), the originating ISUP legs of the call just rack up the toll charges, unaware that the other legs of the call are gone. Bad things also happen if the original INVITE included a Session-Expires: header, which causes the PSTN gateway to emit its own INVITE back to the calling network (via the SER) at intervals during the call, starting at 50% of the time specified in that header. These packets also die a horrible fate in the PSTN-Gateway and SER packet loop, which causes the call to collapse 32 seconds later. (That's unbelieveably bad.) Recognizing this apparent-shortcoming in the pstn.cfg example, and using the fact that the script could determine that this called-party-BYE or INVITE message came from the PSTN gateway and not somewhere else, I added tests to not perform the rewritehost that sends the message back to the gateway, but instead sets the address to that of the known place where the INVITE originally came from, because in this super simple example, there is only one place, eg phone-asterisk-SER-PSTN-SS7network-... Forcing BYE/INVITEs coming from the PSTN direction sort of fixes the problem of called-party BYEs, and the PSTN gateway device emitting re-INVITEs. However, it only works for ONE source, which is lousy as you have to hard-code the right place for the call in both directions! And of course, someone has too many calls for their one SBC (aka an asterisk box) and they have more than one now. In fact, many more, and any of them can send me the same range of calls from the same sources and to the same PSTN gateway. And it would be really nice if I sent the called-party-BYE or re-INVITEs back to the right SBC. But how? An examination of the numerous variables that SER has (as viewed via xlog() when the called-party-BYE or called-network-reINVITE comes in) shows that SER actually has the right IP address available that the called-party-BYE or re-INVITE should be sent back to (apparently saved as part of the transaction details from the original INVITE), but I can't figure out how to reference this value in "rewritehost()". Like most SER functions, rewritehost() doesn't appear to accept variable names as parameters, and as xlog uses cryptic abbreviations for what it displays, I probably wouldn't know what variable name to specify in rewritehost() anyway. (Did two different programmers do these parts?) Now, if you don't use "rewritehost()" in the send-back-to-caller direction and are hoping that SER will do the right thing on its own, you will be disappointed. The t_relay() merely sends the message back itself and you get a much smaller SIP message loop going. (This is because SER puts its IP address all over the original INVITE message so that the PSTN gateway would talk to SER rather than the true source of the call.) So that appears to means that you must set rewritehost() to a reasonable destination to get the message to move beyond SER, and somehow I have to choose a destination from many possible destinations the original INVITE came from, based on some piece of trivia that was in the original INVITE that I can somehow trigger on. That means if there are 250 different SBCs out there sending me calls, and one of them was where the original INVITE came from for the call I just got a called-party-BYE from, how do I force that called-party-BYE (or a called-network re-INVITE) to go back to the right place? Must I build a giant if statement that tests the addresses of all 250 sources and leaves a flag or something that can be used in 250 if statements later when the called-party-BYE or called-network-reINVITE shows up? Sounds horrible and a giant kludge! Any other solutions? Thanks in advance for thinking about this one!

2 1

[Serusers] Is there any way to validate the RTP/media address?
by Frank Durda IV 19 Aug '09

19 Aug '09

I am receiving INVITEs from a specific source, and I have a need to test the media IP address specified in the SDP payload against a list of approved netblocks. I'd like to catch this in the INVITE and reject the call if a RTP address in the SDP payload is out of a pre-defined range for that given source. Is there a way to get this address into a variable that can be used in "if" statements? As in something like this? if (src_ip==17.18.19.20 && method=="INVITE") { if (rtp_addr!=17.0.0.0/8) { sl_reply("502", "Rejected - Unauthorized RTP source"); drop; } } with rtp_addr being the made-up, hope-something-like-it-exists thing. Thanks in advance.

1 0

[Kamailio-Users] Kamailio installed on other dir issues
by Uriel Rozenbaum 18 Aug '09

18 Aug '09

Guys, I've installed all Kamailio in /opt/kamailio. When I execute some kamctl using cr module I get the following: /opt/kamailio/sbin/kamctl cr reload ERROR: This command requires a database engine - none was loaded Do I have to configure something? If i just run /opt/kamailio/sbin/kamctl I get the help lines and running /opt/kamailio/sbin/kamctl fifo cr_reload_routes makes effect as well. Thanks! Uriel

2 4

[Serusers] new_t: out of mem
by Roger Tan 18 Aug '09

18 Aug '09

hi. I tested my ser server which version is 0.9.7+cvs20080424 used sipps sending 1000 invite per sec. after about 10 sec, there was an out of mem error. Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: new_t: out of mem: Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: t_newtran: new_t failed Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: sip_msg_cloner: cannot allocate memory and my memlog is: Aug 18 10:11:02 ubuntu /sbin/ser[20787]: ----------------------------- Aug 18 10:11:02 ubuntu /sbin/ser[20787]: Memory status (shm): Aug 18 10:11:02 ubuntu /sbin/ser[20787]: qm_status (0xb5acf000): Aug 18 10:11:02 ubuntu /sbin/ser[20787]: heap size= 33554432 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: used= 1914592, used+overhead=2224736, free=31329696 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: max used (+overhead)= 33534720 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: dumping all alloc'ed. fragments: Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 0. N address=0xb5ad41a8 frag=0xb5ad41a0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 1. N address=0xb5ad41c8 frag=0xb5ad41c0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 2. N address=0xb5ad41e8 frag=0xb5ad41e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 3. N address=0xb5ad4208 frag=0xb5ad4200 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 4. N address=0xb5ad4228 frag=0xb5ad4220 size=4096 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 5. N address=0xb5ad5238 frag=0xb5ad5230 size=4096 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 6. N address=0xb5ad6248 frag=0xb5ad6240 size=2800 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 7. N address=0xb5ad6d48 frag=0xb5ad6d40 size=1680 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 8. N address=0xb5ad73e8 frag=0xb5ad73e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 9. N address=0xb5ad7408 frag=0xb5ad7400 size=1572864 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 10. N address=0xb5c57418 frag=0xb5c57410 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 11. N address=0xb5c57438 frag=0xb5c57430 size=320 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 12. N address=0xb5c57588 frag=0xb5c57580 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 13. N address=0xb5c575c8 frag=0xb5c575c0 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 14. N address=0xb5c57628 frag=0xb5c57620 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 15. N address=0xb5c57688 frag=0xb5c57680 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 16. N address=0xb5c576e8 frag=0xb5c576e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 17. N address=0xb5c57708 frag=0xb5c57700 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 18. N address=0xb5c57728 frag=0xb5c57720 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 19. N address=0xb5c57748 frag=0xb5c57740 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 20. N address=0xb5c57768 frag=0xb5c57760 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 21. N address=0xb5c57798 frag=0xb5c57790 size=131072 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 22. N address=0xb5c777a8 frag=0xb5c777a0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 23. N address=0xb5c777c8 frag=0xb5c777c0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 24. N address=0xb5c777e8 frag=0xb5c777e0 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 25. N address=0xb5c77818 frag=0xb5c77810 size=131072 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 60. N address=0xb5ca7948 frag=0xb5ca7940 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 61. N address=0xb5ca7988 frag=0xb5ca7980 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 62. N address=0xb5ca79a8 frag=0xb5ca79a0 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 63. N address=0xb5ca7a08 frag=0xb5ca7a00 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 64. N address=0xb5ca7a48 frag=0xb5ca7a40 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 65. N address=0xb5ca7a78 frag=0xb5ca7a70 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 66. N address=0xb5ca7a98 frag=0xb5ca7a90 size=65808 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: dumping free list stats : Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 1. fragments no.: 1232, unused: 0 ^I^I bucket size: 16 - 16 (first 16) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 38. fragments no.: 54, unused: 0 ^I^I bucket size: 608 - 608 (first 608) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 43. fragments no.: 54, unused: 0 ^I^I bucket size: 688 - 688 (first 688) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 44. fragments no.: 3991, unused: 0 ^I^I bucket size: 704 - 704 (first 704) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 57. fragments no.: 114, unused: 0 ^I^I bucket size: 912 - 912 (first 912) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 58. fragments no.: 4004, unused: 0 ^I^I bucket size: 928 - 928 (first 928) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 68. fragments no.: 46, unused: 0 ^I^I bucket size: 1088 - 1088 (first 1088) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 82. fragments no.: 109, unused: 0 ^I^I bucket size: 1312 - 1312 (first 1312) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 115. fragments no.: 34, unused: 0 ^I^I bucket size: 1840 - 1840 (first 1840) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 174. fragments no.: 3795, unused: 0 ^I^I bucket size: 2784 - 2784 (first 2784) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 186. fragments no.: 790, unused: 0 ^I^I bucket size: 2976 - 2976 (first 2976) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 187. fragments no.: 3820, unused: 0 ^I^I bucket size: 2992 - 2992 (first 2992) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 262. fragments no.: 1, unused: 0 ^I^I bucket size: 4192 - 4192 (first 4192) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 318. fragments no.: 1, unused: 0 ^I^I bucket size: 5088 - 5088 (first 5088) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: ----------------------------- I think it's not a memory leak but the so many small memory fragments, and those fragments were not combinated. How I solve this problem? Thanks. -- MSN: miao__tan(a)hotmail.com Email: m.roger.t(a)gmail.com

1 0

[Serusers] tcpconn_new: mem. allocation failure
by Roger Tan 18 Aug '09

18 Aug '09

Hi:

1 0

[Kamailio-Users] Generating SIP /TLS Certificate
by Meftah Tayeb 17 Aug '09

17 Aug '09

hello all please how i can generate a certificate to do sip / tls? and how to configure my kamailio? try to help me please thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4341 (20090817) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

2 1

Re: [Kamailio-Users] presence_dialoginfo context
by David 17 Aug '09

17 Aug '09

Hey, I definitely have an exit(), but even if I didn't most of the message is changed, an exit would not change that would it? David Klaus Darilion wrote: > Very strange. > > Maybe exit(); is missing after t_relay() loop and both PUBLISHs are > processed? I think I forgot the exit() in the config snippet. > > Yes, on the server. And please attach the trace as attachment - inline > is difficult to read > > klaus > > David schrieb: >> Hey, >> >> Here is an expert from the /var/log/messsages : >> >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: New request - >> M=PUBLISH RURI=sip:group1.105@mydomain.tld >> F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld >> IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: Publish >> rewritten forwarded for ext=group1.105 >> ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml >> version="1.0"?> <dialog-info >> xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" >> entity="sip:group1.105@mydomain.tld"> <dialog >> id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" >> call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" >> local-tag="4fbrxu9548" remote-tag="as4fb63283" >> direction="recipient"> <state>early</state> <remote> >> <identity>sip:103@mydomain.tld</identity> <target >> uri="sip:103@mydomain.tld"/> </remote> <local> >> <identity>sip:group1.105@mydomain.tld</identity> <target >> uri="sip:group1.105@mydomain.tld"/> </local> </dialog> >> </dialog-info> >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: New request - >> M=PUBLISH RURI=sip:group1.105@mydomain.tld >> F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld >> IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: Publish >> accepted for ext=group1.105 ruri=sip:group1.105@mydomain.tld >> duri=<null> body=<?xml version="1.0"?> <dialog-info >> xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" >> entity="sip:105@mydomain.tld"> <dialog >> id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" >> call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" >> local-tag="4fbrxu9548" remote-tag="as4fb63283" >> direction="recipient"> <state>early</state> <remote> >> <identity>sip:103@mydomain.tld</identity> <target >> uri="sip:103@mydomain.tld"/> </remote> <local> >> <identity>sip:105@mydomain.tld</identity> <target >> uri="sip:105@mydomain.tld"/> </local> </dialog> </dialog-info> >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: >> INFO:presence:send_notify_request: NOTIFY sip:group1.101@mydomain.tld >> via sip:group1.101@phone.ip:57234 on behalf of sip:105@mydomain.tld >> for event dialog >> >> But the phone's syslog shows this : >> >> Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 >> Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 >> Aug 14 10:16:21 linksys NOTIFY sip:group1.101@phone.ip:57234 >> SIP/2.0^M Via: SIP/2.0/UDP >> 72.55.182.125;branch=z9hG4bK0b0a.9725e23.0^M To: >> sip:group1.101@mydomain.tld;tag=1c32db3af4d889ac^M From: >> sip:105@mydomain.tld;tag=6d6077e15fecee48b2721a5e94bf8883-91a8^M >> CSeq: 12 NOTIFY^M Call-ID: 984d4cd2-37b04f4(a)192.168.1.104^M >> Content-Length: 655^M User-Agent: Kamailio (1.5.2-tls (i386/linux))^M >> Max-Forwards: 70^M Event: dialog^M Contact: <sip:mydomain.tld:5060>^M >> Subscription-State: active;expires=1470^M Content-Type: >> application/dialog-info+xml^M ^M <?xml version="1.0"?> <dialog-info >> xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" >> state="full" entity="group1.105(a)mydomain.tld"> <dialog >> id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" >> call-id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" >> local-tag="loto7h3i9a" remote-tag="as4f284f05" >> direction="recipient"> <state>early</state> <remote> >> <identity>sip:103@mydomain.tld</identity> <target uri=" >> >> Everything is good, but notice the entity is somehow back to group1.105. >> >> Where do you want me to run the traceroute? On the server? >> >> David >> >> Klaus Darilion wrote: >>> >>> >>> David schrieb: >>>> Hey, >>>> >>>> I tried that, but it did not seem to change anything. I saw the >>>> message change on the second loop, but it seems that the presence >>>> module figure out my tactic and fixed the message. >>> >>> That's not possible. The presence module does not alter the received >>> body. I think the problem is somewhere else. >>> >>> Can you send me a trace? >>> >>> ngrep -t -P "" -W byline -d any port 5060 >>> >>> regards >>> klaus >>> >>> >>>> >>>> David >>>> >>>> Klaus Darilion wrote: >>>>> That does not work: presence_server always uses the original >>>>> received body, the modifications applied by you are not seen by >>>>> the presence server. >>>>> >>>>> The workaround would be to "loop" the modified PUBLISH to Kamailio >>>>> again, e.g.: >>>>> >>>>> listen=.......:5060 >>>>> listen=.......:5050 >>>>> >>>>> route{ >>>>> ... >>>>> if (is_method("PUBLISH")) { >>>>> if (dst_port==5060) { >>>>> subst_body('/group1[.]//'); >>>>> t_relay("udp:127.0.0.1:5050); >>>>> } else if (dst_port==5050) { >>>>> handle_publish(); >>>>> t_release(); >>>>> } >>>>> } >>>>> ... >>>>> } >>>>> >>>>> >>>>> klaus >>>>> David schrieb: >>>>>> Hey, >>>>>> >>>>>> I tried doing a subst or replace_body_all to change the entity="" >>>>>> value, >>>>>> >>>>>> if (is_method("PUBLISH") ) >>>>>> { >>>>>> subst_body('/group1[.]//'); >>>>>> handle_publish(); >>>>>> t_release(); >>>>>> } >>>>>> >>>>>> It did not work, the $rb variable still shows the old body and >>>>>> the NOTIFY received on the phone is the old body as well. >>>>>> >>>>>> I thought about doing a string replace on the outbound NOTIFY but >>>>>> it never hits my routing script, so I can't. >>>>>> >>>>>> I read textops and it does not list any limitations relating to me. >>>>>> >>>>>> Any ideas? >>>>>> >>>>>> David >>>>>> >>>>>> Klaus Darilion wrote: >>>>>>> I think the problem lies within the dialog module. Dialog module >>>>>>> stores from/to URI. Both, from and to URI are derived from the >>>>>>> headers. We could extend the dialog module to store request-URI >>>>>>> in the dialog-info structure too, and add an option to >>>>>>> pua_dialoginfo to use RURI instead of To-URI. >>>>>>> >>>>>>> regards >>>>>>> klaus >>>>>>> >>>>>>> David schrieb: >>>>>>>> Hey, >>>>>>>> >>>>>>>> I think I am doing something wrong, because here is the NOTIFY >>>>>>>> that comes to the phone : >>>>>>>> >>>>>>>> NOTIFY sip:group1.101@my.home.ip:57234 SIP/2.0 >>>>>>>> Via: SIP/2.0/UDP my.kamailio.ip;branch=z9hG4bK3cab.78e494a4.0 >>>>>>>> To: sip:group1.101@my.kamailio.domain.name;tag=b2514519391e8b4d >>>>>>>> From: >>>>>>>> sip:102@my.kamailio.domain.name;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>>>> >>>>>>>> CSeq: 12 NOTIFY^M Call-ID: 24db3209-611c031d(a)192.168.1.104 >>>>>>>> Content-Length: 298 >>>>>>>> User-Agent: MyServer 1.0 >>>>>>>> Max-Forwards: 70 >>>>>>>> Event: dialog >>>>>>>> Contact: <sip:my.kamailio.domain.name:5060> >>>>>>>> Subscription-State: active;expires=1570 >>>>>>>> Content-Type: application/dialog-info+xml >>>>>>>> >>>>>>>> <?xml version="1.0"?> <dialog-info >>>>>>>> xmlns="urn:ietf:params:xml:ns:dialog-info" >>>>>>>> version="11" state="full" >>>>>>>> entity="group1.102(a)my.kamailio.domain.name"> <dialog >>>>>>>> id="7aaa32e84362a246716009175ad670be(a)domain.tld" >>>>>>>> direction="recipient"> <state>early</state> </dialog> >>>>>>>> </dialog-info> >>>>>>>> >>>>>>>> The phone ( I tested with a GXP2000, GXP2020 and SPA962 + >>>>>>>> SPA932) does not flash lights or anything. Since you suggested >>>>>>>> that a solid kamailio would work out of the box, I suspect that >>>>>>>> either I miscommunicated my setup or did something really >>>>>>>> wrong. The notify definitely gets to the phone and the phone >>>>>>>> replies 200/OK when it receives the NOTIFY. But I think that >>>>>>>> the telephone is not understanding the request because it >>>>>>>> subscribed to '102' but received a notification for >>>>>>>> 'group1.102'. The funny thing is, the From header matches the >>>>>>>> subscribe To header, it's just the XML has the full name >>>>>>>> instead of the shortened name. >>>>>>>> >>>>>>>> To: >>>>>>>> <sip:102@my.kamailio.domain.name>;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>>>> >>>>>>>> >>>>>>>> I see this in my ( on kamailio ) /var/log/messages : >>>>>>>> >>>>>>>> Aug 13 09:55:24 kamailio-dev /usr/sbin/kamailio[25449]: >>>>>>>> INFO:presence:send_notify_request: NOTIFY >>>>>>>> sip:group1.101@my.kamailio.domain.name via >>>>>>>> sip:group1.101@my.home.ip.where.phone.is:57234 on behalf of >>>>>>>> sip:102@my.kamailio.domain.name for event dialog >>>>>>>> >>>>>>>> I should also mention that the NOTIFY sent out by presence >>>>>>>> bypasses my routing scripts. So I have the PUBLISH come through >>>>>>>> ( which I leave alone ) and the NOTIFY is sent according to the >>>>>>>> location table without ever consulting my routing script. >>>>>>>> >>>>>>>> So everything amazingly worked out well, except that the lights >>>>>>>> are not changing status which I think is related to the XML >>>>>>>> document dialog-info entity attribute containing the group >>>>>>>> name, sent from my server. >>>>>>>> >>>>>>>> Any ideas or suggestions ? >>>>>>>> >>>>>>>> Thanks, >>>>>>>> >>>>>>>> David >>>>>>>> >>>>>>>> Other information : >>>>>>>> >>>>>>>> if ( is_method("SUBSCRIBE") ) >>>>>>>> { >>>>>>>> avp_db_query("select groupname from sometable ", >>>>>>>> "$avp(s:groupname)"); >>>>>>>> avp_printf("$ru", >>>>>>>> "sip:$avp(s:zone).$tU@my.kamailio.domain.name"); >>>>>>>> xlog("L_INFO", "Subscribe rewritten from $tU to >>>>>>>> $ru - M=$rm \n"); >>>>>>>> xlog("L_INFO", "Handling SUBSCRIBE - $fU - >>>>>>>> $avp(s:zone) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >>>>>>>> handle_subscribe(); >>>>>>>> t_release(); >>>>>>>> exit; >>>>>>>> } >>>>>>>> >>>>>>>> Kamailio compiled from sources : >>>>>>>> >>>>>>>> Path: /usr/src/kamailio >>>>>>>> URL: >>>>>>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 >>>>>>>> Repository Root: >>>>>>>> https://openser.svn.sourceforge.net/svnroot/openser >>>>>>>> Repository UUID: 689a6050-402a-0410-94f2-e92a70836424 >>>>>>>> Revision: 5910 >>>>>>>> Node Kind: directory >>>>>>>> Schedule: normal >>>>>>>> Last Changed Author: henningw >>>>>>>> Last Changed Rev: 5910 >>>>>>>> Last Changed Date: 2009-08-06 13:08:30 -0400 (Thu, 06 Aug 2009) >>>>>>>> >>>>>>>> >>>>>>>> Klaus Darilion wrote: >>>>>>>>> notifies are generated by the presence module, based on the >>>>>>>>> subscription. So, there is nothing you have to do with NOTIFY. >>>>>>>>> Also PUBLISHs are created internally. Depending on your >>>>>>>>> concrete setup (e.g. format of the SIP user names) it should >>>>>>>>> work out of the box. >>>>>>>>> >>>>>>>>> klaus >>>>>>>>> >>>>>>>>> dlublink schrieb: >>>>>>>>>> What about the notifies? >>>>>>>>>> >>>>>>>>>> Klaus Darilion wrote: >>>>>>>>>>> dlublink wrote: >>>>>>>>>>>> Hello, >>>>>>>>>>>> >>>>>>>>>>>> I have three different groups of extensions on my kamailio >>>>>>>>>>>> I want to be able to separate them, so I prefixed a name to >>>>>>>>>>>> the extensions, so I have : >>>>>>>>>>>> >>>>>>>>>>>> 1. group1.101 >>>>>>>>>>>> 2. group1.102 >>>>>>>>>>>> 3. group2.101 >>>>>>>>>>>> 4. group2.102 >>>>>>>>>>>> 5. group3.102 >>>>>>>>>>>> 6. group3.103. >>>>>>>>>>>> >>>>>>>>>>>> The phones from different groups can not call each other, I >>>>>>>>>>>> found a pseudo variable that I use to rewrite the >>>>>>>>>>>> destination url, so if user group1.101 dials 102 I rewrite >>>>>>>>>>>> it to group1.102. >>>>>>>>>>>> >>>>>>>>>>>> I want to do the same thing for presence_dialog info, how >>>>>>>>>>>> can I rewrite the subscribe, presence and and notify >>>>>>>>>>>> messages to append the appropriate prefix ? >>>>>>>>>>> >>>>>>>>>>> Just apply the same rewrite you have already done for the >>>>>>>>>>> INVITE also for the SUBSCRIBE >>>>>>>>>>> >>>>>>>>>>> regards >>>>>>>>>>> klaus >>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Thanks, >>>>>>>>>>>> >>>>>>>>>>>> David >>>>>>>>>>>> >>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>> Kamailio (OpenSER) - Users mailing list >>>>>>>>>>>> Users(a)lists.kamailio.org >>>>>>>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>>>>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>>>>>>>>> >>>>>>>>>> >>>>>>>> >>>>>> >>>> >>

2 1

[Serusers] 200 OK not being ACKnowledged
by Steven C. Blair 14 Aug '09

14 Aug '09

I'm testing SIP trunking connectivity to an ITSP through a Covergence (CXC)/ACME Packet session border controller. Calls from an IP phone to the PSTN through SER and the CXC are being dropped 15 seconds after the initial call is setup and just after a cal l is answered. We do not have this problem with calls which reach the PSTN via local gateways. It seems that SER is not relaying an ACK which it receives from the phone which causes the SBC to resend a 200 OK message until finally the connection times out. I've checked the call routing and do not see any issue however I do notice the following in the SER logs. Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: --ACK--: Time: [Thu Aug 13 13:07:24 2009] From: <sip:89390@vidar.net.isc.upenn.edu> To <sip:92155738396@vidar.net.isc.upenn.edu;user=phone> Src IP <128.91.2.223> Method:<ACK> Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: [Thu Aug 13 13:07:24 2009] [453a4bbf-90848c86-ea3a8091(a)128.91.56.18] <sip:92155738396@vidar.net.isc.upenn.edu:5060;user=phone;maddr=128.91.56.242;transport=udp> Starting PRESENCE/BLA section Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: [Thu Aug 13 13:07:24 2009] [453a4bbf-90848c86-ea3a8091(a)128.91.56.18] <sip:92155738396@vidar.net.isc.upenn.edu:5060;user=phone;maddr=128.91.56.242;transport=udp> End PRESENCE/BLA section Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: [Thu Aug 13 13:07:24 2009] [453a4bbf-90848c86-ea3a8091(a)128.91.56.18] --URI match--. Method: <ACK> R-uri: <sip:92155738396@vidar.net.isc.upenn.edu:5060;user=phone;maddr=128.91.56.242;transport=udp> Contact Header: <<sip:89390@128.91.56.18:5060>> From: <sip:89390@vidar.net.isc.upenn.edu> To <sip:92155738396@vidar.net.isc.upenn.edu;user=phone> IP source address <128.91.2.223> Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: --ACK--: Time: [Thu Aug 13 13:07:24 2009] From: <sip:89390@vidar.net.isc.upenn.edu> To <sip:92155738396@vidar.net.isc.upenn.edu;user=phone> Src IP <128.91.2.223> Method:<ACK> Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: Too many hops Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: Warning: sl_send_reply: I won't send a reply for ACK!! User 89390 is a local subscriber and the phone initiating the call. The recipient of the call is my desk phone 215-573-8396 which is on a different, unrelated SER service. We do record_route traffic through SER for call control and accounting reasons. It seems like the ACK is looping through SER and therefore never being sent to the SBC but I cannot see why. Does anyone have any thoughts on this issue? Thanks,Steve

3 5

Re: [Kamailio-Users] presence_dialoginfo context
by David 14 Aug '09

14 Aug '09

Hey, Apparently the list did not accept the previous message... Here is the code you suggested with my modifications, it'll help put the log extracts in context : else if ( dst_port == 5060 ) { xlog("L_INFO", "Publish rewritten forwarded for ext=$tU ruri=$ru duri=$du body=$rb"); subst_body('/group1[.]//g' ) ; t_relay("udp:127.0.0.1:5050"); exit ; } else { xlog("L_INFO", "Publish accepted for ext=$tU ruri=$ru duri=$du body=$rb"); handle_publish(); t_release(); exit; } Here is an expert from the /var/log/messsages : Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: New request - M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: Publish rewritten forwarded for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:group1.105@mydomain.tld"> <dialog id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" local-tag="4fbrxu9548" remote-tag="as4fb63283" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri="sip:103@mydomain.tld"/> </remote> <local> <identity>sip:group1.105@mydomain.tld</identity> <target uri="sip:group1.105@mydomain.tld"/> </local> </dialog> </dialog-info> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: New request - M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: Publish accepted for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:105@mydomain.tld"> <dialog id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" local-tag="4fbrxu9548" remote-tag="as4fb63283" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri="sip:103@mydomain.tld"/> </remote> <local> <identity>sip:105@mydomain.tld</identity> <target uri="sip:105@mydomain.tld"/> </local> </dialog> </dialog-info> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: INFO:presence:send_notify_request: NOTIFY sip:group1.101@mydomain.tld via sip:group1.101@phone.ip:57234 on behalf of sip:105@mydomain.tld for event dialog But the phone's syslog shows this : Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 Aug 14 10:16:21 linksys NOTIFY sip:group1.101@phone.ip:57234 SIP/2.0^M Via: SIP/2.0/UDP 72.55.182.125;branch=z9hG4bK0b0a.9725e23.0^M To: sip:group1.101@mydomain.tld;tag=1c32db3af4d889ac^M From: sip:105@mydomain.tld;tag=6d6077e15fecee48b2721a5e94bf8883-91a8^M CSeq: 12 NOTIFY^M Call-ID: 984d4cd2-37b04f4(a)192.168.1.104^M Content-Length: 655^M User-Agent: Kamailio (1.5.2-tls (i386/linux))^M Max-Forwards: 70^M Event: dialog^M Contact: <sip:mydomain.tld:5060>^M Subscription-State: active;expires=1470^M Content-Type: application/dialog-info+xml^M ^M <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="group1.105(a)mydomain.tld"> <dialog id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" call-id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" local-tag="loto7h3i9a" remote-tag="as4f284f05" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri=" Everything is good, but notice the entity is somehow back to group1.105. Where do you want me to run the traceroute? On the server? David Klaus Darilion wrote: > > > David schrieb: >> Hey, >> >> I tried that, but it did not seem to change anything. I saw the >> message change on the second loop, but it seems that the presence >> module figure out my tactic and fixed the message. > > That's not possible. The presence module does not alter the received > body. I think the problem is somewhere else. > > Can you send me a trace? > > ngrep -t -P "" -W byline -d any port 5060 > > regards > klaus > > >> >> David >> >> Klaus Darilion wrote: >>> That does not work: presence_server always uses the original >>> received body, the modifications applied by you are not seen by the >>> presence server. >>> >>> The workaround would be to "loop" the modified PUBLISH to Kamailio >>> again, e.g.: >>> >>> listen=.......:5060 >>> listen=.......:5050 >>> >>> route{ >>> ... >>> if (is_method("PUBLISH")) { >>> if (dst_port==5060) { >>> subst_body('/group1[.]//'); >>> t_relay("udp:127.0.0.1:5050); >>> } else if (dst_port==5050) { >>> handle_publish(); >>> t_release(); >>> } >>> } >>> ... >>> } >>> >>> >>> klaus >>> David schrieb: >>>> Hey, >>>> >>>> I tried doing a subst or replace_body_all to change the entity="" >>>> value, >>>> >>>> if (is_method("PUBLISH") ) >>>> { >>>> subst_body('/group1[.]//'); >>>> handle_publish(); >>>> t_release(); >>>> } >>>> >>>> It did not work, the $rb variable still shows the old body and the >>>> NOTIFY received on the phone is the old body as well. >>>> >>>> I thought about doing a string replace on the outbound NOTIFY but >>>> it never hits my routing script, so I can't. >>>> >>>> I read textops and it does not list any limitations relating to me. >>>> >>>> Any ideas? >>>> >>>> David >>>> >>>> Klaus Darilion wrote: >>>>> I think the problem lies within the dialog module. Dialog module >>>>> stores from/to URI. Both, from and to URI are derived from the >>>>> headers. We could extend the dialog module to store request-URI in >>>>> the dialog-info structure too, and add an option to pua_dialoginfo >>>>> to use RURI instead of To-URI. >>>>> >>>>> regards >>>>> klaus >>>>> >>>>> David schrieb: >>>>>> Hey, >>>>>> >>>>>> I think I am doing something wrong, because here is the NOTIFY >>>>>> that comes to the phone : >>>>>> >>>>>> NOTIFY sip:group1.101@my.home.ip:57234 SIP/2.0 >>>>>> Via: SIP/2.0/UDP my.kamailio.ip;branch=z9hG4bK3cab.78e494a4.0 >>>>>> To: sip:group1.101@my.kamailio.domain.name;tag=b2514519391e8b4d >>>>>> From: >>>>>> sip:102@my.kamailio.domain.name;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>> >>>>>> CSeq: 12 NOTIFY^M Call-ID: 24db3209-611c031d(a)192.168.1.104 >>>>>> Content-Length: 298 >>>>>> User-Agent: MyServer 1.0 >>>>>> Max-Forwards: 70 >>>>>> Event: dialog >>>>>> Contact: <sip:my.kamailio.domain.name:5060> >>>>>> Subscription-State: active;expires=1570 >>>>>> Content-Type: application/dialog-info+xml >>>>>> >>>>>> <?xml version="1.0"?> <dialog-info >>>>>> xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" >>>>>> state="full" entity="group1.102(a)my.kamailio.domain.name"> >>>>>> <dialog id="7aaa32e84362a246716009175ad670be(a)domain.tld" >>>>>> direction="recipient"> <state>early</state> </dialog> >>>>>> </dialog-info> >>>>>> >>>>>> The phone ( I tested with a GXP2000, GXP2020 and SPA962 + SPA932) >>>>>> does not flash lights or anything. Since you suggested that a >>>>>> solid kamailio would work out of the box, I suspect that either I >>>>>> miscommunicated my setup or did something really wrong. The >>>>>> notify definitely gets to the phone and the phone replies 200/OK >>>>>> when it receives the NOTIFY. But I think that the telephone is >>>>>> not understanding the request because it subscribed to '102' but >>>>>> received a notification for 'group1.102'. The funny thing is, the >>>>>> From header matches the subscribe To header, it's just the XML >>>>>> has the full name instead of the shortened name. >>>>>> >>>>>> To: >>>>>> <sip:102@my.kamailio.domain.name>;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>> >>>>>> >>>>>> I see this in my ( on kamailio ) /var/log/messages : >>>>>> >>>>>> Aug 13 09:55:24 kamailio-dev /usr/sbin/kamailio[25449]: >>>>>> INFO:presence:send_notify_request: NOTIFY >>>>>> sip:group1.101@my.kamailio.domain.name via >>>>>> sip:group1.101@my.home.ip.where.phone.is:57234 on behalf of >>>>>> sip:102@my.kamailio.domain.name for event dialog >>>>>> >>>>>> I should also mention that the NOTIFY sent out by presence >>>>>> bypasses my routing scripts. So I have the PUBLISH come through ( >>>>>> which I leave alone ) and the NOTIFY is sent according to the >>>>>> location table without ever consulting my routing script. >>>>>> >>>>>> So everything amazingly worked out well, except that the lights >>>>>> are not changing status which I think is related to the XML >>>>>> document dialog-info entity attribute containing the group name, >>>>>> sent from my server. >>>>>> >>>>>> Any ideas or suggestions ? >>>>>> >>>>>> Thanks, >>>>>> >>>>>> David >>>>>> >>>>>> Other information : >>>>>> >>>>>> if ( is_method("SUBSCRIBE") ) >>>>>> { >>>>>> avp_db_query("select groupname from sometable ", >>>>>> "$avp(s:groupname)"); >>>>>> avp_printf("$ru", >>>>>> "sip:$avp(s:zone).$tU@my.kamailio.domain.name"); >>>>>> xlog("L_INFO", "Subscribe rewritten from $tU to $ru >>>>>> - M=$rm \n"); >>>>>> xlog("L_INFO", "Handling SUBSCRIBE - $fU - >>>>>> $avp(s:zone) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >>>>>> handle_subscribe(); >>>>>> t_release(); >>>>>> exit; >>>>>> } >>>>>> >>>>>> Kamailio compiled from sources : >>>>>> >>>>>> Path: /usr/src/kamailio >>>>>> URL: >>>>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 >>>>>> Repository Root: https://openser.svn.sourceforge.net/svnroot/openser >>>>>> Repository UUID: 689a6050-402a-0410-94f2-e92a70836424 >>>>>> Revision: 5910 >>>>>> Node Kind: directory >>>>>> Schedule: normal >>>>>> Last Changed Author: henningw >>>>>> Last Changed Rev: 5910 >>>>>> Last Changed Date: 2009-08-06 13:08:30 -0400 (Thu, 06 Aug 2009) >>>>>> >>>>>> >>>>>> Klaus Darilion wrote: >>>>>>> notifies are generated by the presence module, based on the >>>>>>> subscription. So, there is nothing you have to do with NOTIFY. >>>>>>> Also PUBLISHs are created internally. Depending on your concrete >>>>>>> setup (e.g. format of the SIP user names) it should work out of >>>>>>> the box. >>>>>>> >>>>>>> klaus >>>>>>> >>>>>>> dlublink schrieb: >>>>>>>> What about the notifies? >>>>>>>> >>>>>>>> Klaus Darilion wrote: >>>>>>>>> dlublink wrote: >>>>>>>>>> Hello, >>>>>>>>>> >>>>>>>>>> I have three different groups of extensions on my kamailio I >>>>>>>>>> want to be able to separate them, so I prefixed a name to the >>>>>>>>>> extensions, so I have : >>>>>>>>>> >>>>>>>>>> 1. group1.101 >>>>>>>>>> 2. group1.102 >>>>>>>>>> 3. group2.101 >>>>>>>>>> 4. group2.102 >>>>>>>>>> 5. group3.102 >>>>>>>>>> 6. group3.103. >>>>>>>>>> >>>>>>>>>> The phones from different groups can not call each other, I >>>>>>>>>> found a pseudo variable that I use to rewrite the destination >>>>>>>>>> url, so if user group1.101 dials 102 I rewrite it to group1.102. >>>>>>>>>> >>>>>>>>>> I want to do the same thing for presence_dialog info, how can >>>>>>>>>> I rewrite the subscribe, presence and and notify messages to >>>>>>>>>> append the appropriate prefix ? >>>>>>>>> >>>>>>>>> Just apply the same rewrite you have already done for the >>>>>>>>> INVITE also for the SUBSCRIBE >>>>>>>>> >>>>>>>>> regards >>>>>>>>> klaus >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Thanks, >>>>>>>>>> >>>>>>>>>> David >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (OpenSER) - Users mailing list >>>>>>>>>> Users(a)lists.kamailio.org >>>>>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>>>>> >>>>>> >>>> >>

1 0

Re: [Kamailio-Users] presence_dialoginfo context
by David 14 Aug '09

14 Aug '09

Hey, Here is an expert from the /var/log/messsages : Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: New request - M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: Publish rewritten forwarded for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:group1.105@mydomain.tld"> <dialog id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" local-tag="4fbrxu9548" remote-tag="as4fb63283" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri="sip:103@mydomain.tld"/> </remote> <local> <identity>sip:group1.105@mydomain.tld</identity> <target uri="sip:group1.105@mydomain.tld"/> </local> </dialog> </dialog-info> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: New request - M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: Publish accepted for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:105@mydomain.tld"> <dialog id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" local-tag="4fbrxu9548" remote-tag="as4fb63283" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri="sip:103@mydomain.tld"/> </remote> <local> <identity>sip:105@mydomain.tld</identity> <target uri="sip:105@mydomain.tld"/> </local> </dialog> </dialog-info> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: INFO:presence:send_notify_request: NOTIFY sip:group1.101@mydomain.tld via sip:group1.101@phone.ip:57234 on behalf of sip:105@mydomain.tld for event dialog But the phone's syslog shows this : Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 Aug 14 10:16:21 linksys NOTIFY sip:group1.101@phone.ip:57234 SIP/2.0^M Via: SIP/2.0/UDP 72.55.182.125;branch=z9hG4bK0b0a.9725e23.0^M To: sip:group1.101@mydomain.tld;tag=1c32db3af4d889ac^M From: sip:105@mydomain.tld;tag=6d6077e15fecee48b2721a5e94bf8883-91a8^M CSeq: 12 NOTIFY^M Call-ID: 984d4cd2-37b04f4(a)192.168.1.104^M Content-Length: 655^M User-Agent: Kamailio (1.5.2-tls (i386/linux))^M Max-Forwards: 70^M Event: dialog^M Contact: <sip:mydomain.tld:5060>^M Subscription-State: active;expires=1470^M Content-Type: application/dialog-info+xml^M ^M <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="group1.105(a)mydomain.tld"> <dialog id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" call-id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" local-tag="loto7h3i9a" remote-tag="as4f284f05" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri=" Everything is good, but notice the entity is somehow back to group1.105. Where do you want me to run the traceroute? On the server? David Klaus Darilion wrote: > > > David schrieb: >> Hey, >> >> I tried that, but it did not seem to change anything. I saw the >> message change on the second loop, but it seems that the presence >> module figure out my tactic and fixed the message. > > That's not possible. The presence module does not alter the received > body. I think the problem is somewhere else. > > Can you send me a trace? > > ngrep -t -P "" -W byline -d any port 5060 > > regards > klaus > > >> >> David >> >> Klaus Darilion wrote: >>> That does not work: presence_server always uses the original >>> received body, the modifications applied by you are not seen by the >>> presence server. >>> >>> The workaround would be to "loop" the modified PUBLISH to Kamailio >>> again, e.g.: >>> >>> listen=.......:5060 >>> listen=.......:5050 >>> >>> route{ >>> ... >>> if (is_method("PUBLISH")) { >>> if (dst_port==5060) { >>> subst_body('/group1[.]//'); >>> t_relay("udp:127.0.0.1:5050); >>> } else if (dst_port==5050) { >>> handle_publish(); >>> t_release(); >>> } >>> } >>> ... >>> } >>> >>> >>> klaus >>> David schrieb: >>>> Hey, >>>> >>>> I tried doing a subst or replace_body_all to change the entity="" >>>> value, >>>> >>>> if (is_method("PUBLISH") ) >>>> { >>>> subst_body('/group1[.]//'); >>>> handle_publish(); >>>> t_release(); >>>> } >>>> >>>> It did not work, the $rb variable still shows the old body and the >>>> NOTIFY received on the phone is the old body as well. >>>> >>>> I thought about doing a string replace on the outbound NOTIFY but >>>> it never hits my routing script, so I can't. >>>> >>>> I read textops and it does not list any limitations relating to me. >>>> >>>> Any ideas? >>>> >>>> David >>>> >>>> Klaus Darilion wrote: >>>>> I think the problem lies within the dialog module. Dialog module >>>>> stores from/to URI. Both, from and to URI are derived from the >>>>> headers. We could extend the dialog module to store request-URI in >>>>> the dialog-info structure too, and add an option to pua_dialoginfo >>>>> to use RURI instead of To-URI. >>>>> >>>>> regards >>>>> klaus >>>>> >>>>> David schrieb: >>>>>> Hey, >>>>>> >>>>>> I think I am doing something wrong, because here is the NOTIFY >>>>>> that comes to the phone : >>>>>> >>>>>> NOTIFY sip:group1.101@my.home.ip:57234 SIP/2.0 >>>>>> Via: SIP/2.0/UDP my.kamailio.ip;branch=z9hG4bK3cab.78e494a4.0 >>>>>> To: sip:group1.101@my.kamailio.domain.name;tag=b2514519391e8b4d >>>>>> From: >>>>>> sip:102@my.kamailio.domain.name;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>> >>>>>> CSeq: 12 NOTIFY^M Call-ID: 24db3209-611c031d(a)192.168.1.104 >>>>>> Content-Length: 298 >>>>>> User-Agent: MyServer 1.0 >>>>>> Max-Forwards: 70 >>>>>> Event: dialog >>>>>> Contact: <sip:my.kamailio.domain.name:5060> >>>>>> Subscription-State: active;expires=1570 >>>>>> Content-Type: application/dialog-info+xml >>>>>> >>>>>> <?xml version="1.0"?> <dialog-info >>>>>> xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" >>>>>> state="full" entity="group1.102(a)my.kamailio.domain.name"> >>>>>> <dialog id="7aaa32e84362a246716009175ad670be(a)domain.tld" >>>>>> direction="recipient"> <state>early</state> </dialog> >>>>>> </dialog-info> >>>>>> >>>>>> The phone ( I tested with a GXP2000, GXP2020 and SPA962 + SPA932) >>>>>> does not flash lights or anything. Since you suggested that a >>>>>> solid kamailio would work out of the box, I suspect that either I >>>>>> miscommunicated my setup or did something really wrong. The >>>>>> notify definitely gets to the phone and the phone replies 200/OK >>>>>> when it receives the NOTIFY. But I think that the telephone is >>>>>> not understanding the request because it subscribed to '102' but >>>>>> received a notification for 'group1.102'. The funny thing is, the >>>>>> From header matches the subscribe To header, it's just the XML >>>>>> has the full name instead of the shortened name. >>>>>> >>>>>> To: >>>>>> <sip:102@my.kamailio.domain.name>;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>> >>>>>> >>>>>> I see this in my ( on kamailio ) /var/log/messages : >>>>>> >>>>>> Aug 13 09:55:24 kamailio-dev /usr/sbin/kamailio[25449]: >>>>>> INFO:presence:send_notify_request: NOTIFY >>>>>> sip:group1.101@my.kamailio.domain.name via >>>>>> sip:group1.101@my.home.ip.where.phone.is:57234 on behalf of >>>>>> sip:102@my.kamailio.domain.name for event dialog >>>>>> >>>>>> I should also mention that the NOTIFY sent out by presence >>>>>> bypasses my routing scripts. So I have the PUBLISH come through ( >>>>>> which I leave alone ) and the NOTIFY is sent according to the >>>>>> location table without ever consulting my routing script. >>>>>> >>>>>> So everything amazingly worked out well, except that the lights >>>>>> are not changing status which I think is related to the XML >>>>>> document dialog-info entity attribute containing the group name, >>>>>> sent from my server. >>>>>> >>>>>> Any ideas or suggestions ? >>>>>> >>>>>> Thanks, >>>>>> >>>>>> David >>>>>> >>>>>> Other information : >>>>>> >>>>>> if ( is_method("SUBSCRIBE") ) >>>>>> { >>>>>> avp_db_query("select groupname from sometable ", >>>>>> "$avp(s:groupname)"); >>>>>> avp_printf("$ru", >>>>>> "sip:$avp(s:zone).$tU@my.kamailio.domain.name"); >>>>>> xlog("L_INFO", "Subscribe rewritten from $tU to $ru >>>>>> - M=$rm \n"); >>>>>> xlog("L_INFO", "Handling SUBSCRIBE - $fU - >>>>>> $avp(s:zone) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >>>>>> handle_subscribe(); >>>>>> t_release(); >>>>>> exit; >>>>>> } >>>>>> >>>>>> Kamailio compiled from sources : >>>>>> >>>>>> Path: /usr/src/kamailio >>>>>> URL: >>>>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 >>>>>> Repository Root: https://openser.svn.sourceforge.net/svnroot/openser >>>>>> Repository UUID: 689a6050-402a-0410-94f2-e92a70836424 >>>>>> Revision: 5910 >>>>>> Node Kind: directory >>>>>> Schedule: normal >>>>>> Last Changed Author: henningw >>>>>> Last Changed Rev: 5910 >>>>>> Last Changed Date: 2009-08-06 13:08:30 -0400 (Thu, 06 Aug 2009) >>>>>> >>>>>> >>>>>> Klaus Darilion wrote: >>>>>>> notifies are generated by the presence module, based on the >>>>>>> subscription. So, there is nothing you have to do with NOTIFY. >>>>>>> Also PUBLISHs are created internally. Depending on your concrete >>>>>>> setup (e.g. format of the SIP user names) it should work out of >>>>>>> the box. >>>>>>> >>>>>>> klaus >>>>>>> >>>>>>> dlublink schrieb: >>>>>>>> What about the notifies? >>>>>>>> >>>>>>>> Klaus Darilion wrote: >>>>>>>>> dlublink wrote: >>>>>>>>>> Hello, >>>>>>>>>> >>>>>>>>>> I have three different groups of extensions on my kamailio I >>>>>>>>>> want to be able to separate them, so I prefixed a name to the >>>>>>>>>> extensions, so I have : >>>>>>>>>> >>>>>>>>>> 1. group1.101 >>>>>>>>>> 2. group1.102 >>>>>>>>>> 3. group2.101 >>>>>>>>>> 4. group2.102 >>>>>>>>>> 5. group3.102 >>>>>>>>>> 6. group3.103. >>>>>>>>>> >>>>>>>>>> The phones from different groups can not call each other, I >>>>>>>>>> found a pseudo variable that I use to rewrite the destination >>>>>>>>>> url, so if user group1.101 dials 102 I rewrite it to group1.102. >>>>>>>>>> >>>>>>>>>> I want to do the same thing for presence_dialog info, how can >>>>>>>>>> I rewrite the subscribe, presence and and notify messages to >>>>>>>>>> append the appropriate prefix ? >>>>>>>>> >>>>>>>>> Just apply the same rewrite you have already done for the >>>>>>>>> INVITE also for the SUBSCRIBE >>>>>>>>> >>>>>>>>> regards >>>>>>>>> klaus >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Thanks, >>>>>>>>>> >>>>>>>>>> David >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (OpenSER) - Users mailing list >>>>>>>>>> Users(a)lists.kamailio.org >>>>>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>>>>> >>>>>> >>>> >>

1 0

[Kamailio-Users] Generate INVITE from Openser
by ALAEDDINE abbech 14 Aug '09

14 Aug '09

Hello all How can i generate an INVITE from OpenSER. That means that Openser make the role of UAC. Who have an idea for this pb. Thanks Regards Aladin

6 9

[Kamailio-Users] presence_dialoginfo context
by dlublink 14 Aug '09

14 Aug '09

Hello, I have three different groups of extensions on my kamailio I want to be able to separate them, so I prefixed a name to the extensions, so I have : 1. group1.101 2. group1.102 3. group2.101 4. group2.102 5. group3.102 6. group3.103. The phones from different groups can not call each other, I found a pseudo variable that I use to rewrite the destination url, so if user group1.101 dials 102 I rewrite it to group1.102. I want to do the same thing for presence_dialog info, how can I rewrite the subscribe, presence and and notify messages to append the appropriate prefix ? Thanks, David

4 10

[Kamailio-Users] Load Balancing Asterisk using Kamailio
by Kurt Weasel 14 Aug '09

14 Aug '09

Hi, I searched around the web to load balance asterisk servers and found Kamailio for possible solution. Let's say I have two identical asterisk servers with same dialplan and configuration and I want both servers look like they have same IP address from clients. Based on my understanding, we need 1 Kamailio as load balancer and 2 Asterisk servers as the real servers. Let's say the setup is : Kamailio load balancer 192.168.2.1 Asterisk Server #1 192.168.2.2 Asterisk Server #2 192.168.2.3 My question is, X-Lite softphone Configuration should be set to domain 192.168.2.1, right? I also want to know the step by step configuration to set kamailio as load balancer. I have not used Kamailio before. However after searching the documentation, the step (based on my understanding) is somewhat like this : 1. Install Kamailio. I will use the step by step here : http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn 2. Then, using dipatcher module, I will configure it using this guide here : http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha Then modify the dispatcher.list file to match the IP address of my asterisk servers : *1 sip:192.168.2.2:5060 1 sip:192.168.2.3:5060* Am I missing some steps? Do I also need to configure dialplan or any other file at Kamailio load balancer? Or those two steps basically done it all for simple load balancing configuration? Thanks for your responses. Best Regard, Kurt Weasel.

5 8

[Kamailio-Users] OpenSER without MYSQL
by Ratheendran R 13 Aug '09

13 Aug '09

Hello everyone, I want to port openSER on an embedded board,I need configure the openser without MySQL due to space constraint. So can you let me know whether this is possible also can you give any reference document on how to port on ARM 9 based boad. Regards, Rathndran <users(a)lists.kamailio.org>

7 9

[Kamailio-Users] Fwd: dialog info + Grandstreams = freezing
by Daniel-Constantin Mierla 13 Aug '09

13 Aug '09

.. fwd to list ... reply all button mismatch ... ---------- Forwarded message ---------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Wed, Aug 12, 2009 at 9:20 AM Subject: Re: [Kamailio-Users] dialog info + Grandstreams = freezing To: David <kamailio.org(a)spam.lublink.net> Hello, On Tue, Aug 11, 2009 at 2:18 PM, David <kamailio.org(a)spam.lublink.net>wrote: > Hey, > > I am not good at diagrams, but here goes : > > > [ Asterisk] [ Asterisk ] ================> [ MySQL ] > / \ / \ > [ Kamailio ] ============/\ > / \ / \ / \ > [ Extension ] [ Extension ] [ Extension ] > > Here is a simple flow : > > 1. Extension 1 is plugged in > 2. It registers to Kamailio and Kamailio stores to Location > 3. Extension 1 subscribes to all the other extensions > 4. Kamailio calls handle_publish() and t_release() > 5. Extension 1 dials '102' > 6. Kamailio receives the INVITE chooses an asterisk server and relays the > INVITE ( t_relay ) > 7. Asterisk does some checking, starts monitor ( if needed ) and some other > stuff > 8. Asterisk sends INVITE to Kamailio for testspace.102 > 9. Kamailio finds testspace.102 in database and forwards INVITE to the > appropriate extension > 10. Extension replies 'OK' > 11. Kamailio, using reply-route, sends it back to asterisk > 12. Asterisk bridges the call with 101 > 13. Asterisk sends an OK back through Kamailio to Extension 1. > 14. Asterisk tries to reinvite the audio ( if not monitoring ) > 15. Audio bypasses Asterisk, but the SIP still is making the full trip. > > Does this answer your question? Do you do call tracking with dialog for both legs (inbond: phone to asterisk; outbound: asterisk to phone)? You should do it for one, outbound is the best, in case asterisk auto-answers the inbound leg. Cheers, Daniel > > David > > > > Daniel-Constantin Mierla wrote: > >> Hello, >> >> I am using the module with snom 370 and works ok. The BLF is on for >> Polycoms but they do not implement the call pickup properly (they have >> direct call pickup as far as I could get). >> >> Please take latest kamailio 1.5 from svn branch 1.5 as there were >> committed some updates (maybe they don't affect you, being mainly to rls, >> but is better to have the latest stable so we refer to same source code). >> >> What would be good is to have a diagram of the signaling, since you say >> there are lot of messages sent around. >> http://callflow.sourceforge.net/ >> >> Cheers, >> Daniel >> >> >> On 11.08.2009 5:03 Uhr, David wrote: >> >>> Hello, >>> >>> I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, >>> 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS >>> 1.5.2-tls compilde on Ubuntu 8.04 >>> >>> I am trying to setup presence_dialoginfo with my Grandstreams, Snom and >>> Linksys. I have a 4 phones on the server. >>> >>> 101 - Linksys SPA962 ( 6.1.5a ) >>> 102 - Grandstream GXP2020 ( 1.2.1.4 ) >>> 103 - Grandstream GXP2000 ( 1.2.1.4 ) >>> 104 - Grandstream GXP2000 ( 1.1.6.46 ) >>> 105 - Snom 360 ( 7.3.23 ) >>> >>> My Kamailio deals with registrations, NAT and BLF everything else is sent >>> to one of two asterisk boxes. I use the dispatcher module for this. This >>> means that when I call one extension to the other, both call legs from >>> asterisk are going through Kamailio as separate calls. But to divide my >>> customers, the usernames are different from the URL that the user types. For >>> example the customer dials '101' but it is changed to testspace.101 when it >>> comes back from asterisk. So Kamailio would have two calls in the event that >>> 101 dials 102. >>> >>> sip:testspace.101@myserver to 102 ( this is sent to asterisk ) >>> sip:testspace.101@myserver to testspace.102 ( this is coming back from >>> asterisk ) >>> >>> Something is horribly wrong. I have the following problems : >>> >>> 1. If 102 calls 103, when 103 answers both phones hang for about 2 >>> minutes >>> 2. If 105 calls 101, 101 BLF comes back to the inactive state ( green on >>> the Linksys and dark on the Snom), but the orange light stays on on the Snom >>> and it thinks the call is still active ( the light is on, but the call is >>> over ) >>> 3. If any extension calls any extension and I try a call pickup, it >>> fails. It looks like the Linksys is sending a NOTIFY to pickup the call ( I >>> thought it was supposed to send an invite... ? ) >>> >>> Looking at the logs it looks like Kamailio is sending out so many NOTIFYs >>> that it is crashing the Grandstreams, and causing the Snom to act funny. >>> >>> Here are some experts from my config file : >>> >>> root@kamailio-dev:/etc/kamailio# grep dialog * >>> kamailio.cfg:# * avp value for dialogs is still not correct >>> kamailio.cfg:loadmodule "dialog.so" >>> kamailio.cfg:loadmodule "presence_dialoginfo.so" >>> kamailio.cfg:loadmodule "pua_dialoginfo.so" >>> kamailio.cfg:#modparam("pua_dialoginfo", "include_localremote", 0) >>> kamailio.cfg:#modparam("pua_dialoginfo", "include_tags", 0) >>> kamailio.cfg:#modparam("pua_dialoginfo", "include_callid", 0) >>> kamailio.cfg:modparam("dialog", "dlg_flag", 4) >>> kamailio.cfg:modparam("dialog", "db_mode", 1) >>> kamailio.cfg:modparam("dialog", "timeout_avp", "$avp(i:10)") # I still >>> haven't figured out how to set $avp(i:10) >>> kamailio.cfg:modparam("pua_dialoginfo", "override_lifetime", 300) >>> kamailio.cfg:modparam("presence_dialoginfo", "force_single_dialog", 1) >>> kamailio.cfg:modparam("pua_dialoginfo", "caller_confirmed", 1) >>> kamailio.cfg:modparam("auth_db|usrloc|acc|domain|avpops|presence|presence_xml|pua|dialog", >>> "db_url", >>> kamailio.cfg:# Flag 4 = Mark the current request for a dialog >>> kamailio.cfg: # sequential request withing a dialog should >>> >>> the set flag looks like this : >>> if ( ds_is_from_list() ) >>> { >>> xlog("L_INFO", "Coming from asterisk"); >>> if ( is_method("INVITE")) >>> { >>> setflag(4); >>> } >>> } >>> >>> So the dialog flag is only set for the leg coming back from asterisk. >>> >>> When a notify comes in : >>> >>> if(is_method("NOTIFY") ) >>> { >>> if (! t_newtran()) >>> { >>> sl_reply_error(); >>> exit; >>> }; >>> >>> t_reply("200", "OK"); >>> t_release(); >>> exit ; >>> } >>> >>> Publish and subscribe are like this : >>> >>> >>> if( is_method("PUBLISH") || is_method("SUBSCRIBE") ) >>> { >>> route(5); >>> exit; >>> } >>> >>> route[5] >>> { >>> # absorb retransmissions >>> if (! t_newtran()) >>> { >>> xlog("L_INFO", "Ignoring PUBLISH/SUBSCRIBE on retransmition - >>> M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >>> sl_reply_error(); >>> exit; >>> }; >>> >>> append_to_reply("Contact: <sip:myserver.tld:5060>\r\n"); >>> >>> if(is_method("PUBLISH")) >>> { >>> handle_publish(); >>> t_release(); >>> >>> } else if( is_method("SUBSCRIBE")) { >>> handle_subscribe(); >>> t_release(); >>> } >>> else >>> { >>> } >>> >>> exit; >>> } >>> >>> I also have NAT checking for those telephones where stun isn't enough. >>> Before I reach publish/subscribe/invite/notify, I also call setbflag() and >>> sometimes call fix_nated_contact(). Additionnally, I have a block if code >>> before my presence stuff if ( has_totag() && loose_route()) { t_relay(); }. >>> >>> If sip.conf:canreinvite=yes, the grandstreams freeze so long that the >>> server times out, and the BLFs get really messed up. if sip:canreinvite=no >>> the grandstreams only freeze for about 30 seconds. >>> >>> Obviously I am doing something wrong, but despite having searched google >>> for endless hours, and poured over documentation, I can not seem to find >>> what I did wrong. >>> >>> I would really appreciate if someone could shed light on my problem. >>> >>> I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, >>> 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS >>> 1.5.2-tls compilde on Ubuntu 8.04 >>> >>> Thanks, >>> >>> David >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >> > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] acc doesn't write to database and to a log file
by Koffedor 13 Aug '09

13 Aug '09

Hello to all I enabled the acc module but it isn't writing to "acc" or "missed_calls" table in mysql, and to a log file too. my config: (...) loadmodule "db_mysql.so" loadmodule "tm.so" loadmodule "rr.so" (...) # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://xxx:xxxx@localhost/kamailio") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") Flags 1 and 2 set just like in a sample configuration file. (...) if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); (...) # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } Mysql tables are created: | acc | CREATE TABLE `acc` ( `id` int(10) unsigned NOT NULL auto_increment, `method` varchar(16) NOT NULL default '', `from_tag` varchar(64) NOT NULL default '', `to_tag` varchar(64) NOT NULL default '', `callid` varchar(128) NOT NULL default '', `sip_code` char(3) NOT NULL default '', `sip_reason` varchar(32) NOT NULL default '', `time` datetime NOT NULL default '0000-00-00 00:00:00', `src_ip` varchar(64) NOT NULL default '', `dst_user` varchar(64) NOT NULL default '', `dst_domain` varchar(128) NOT NULL default '', `src_user` varchar(64) NOT NULL default '', `src_domain` varchar(128) NOT NULL default '', `cdr_id` int(11) NOT NULL default '0', `dst_ouser` varchar(64) NOT NULL default '', PRIMARY KEY (`id`), KEY `acc_callid` (`callid`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 | | missed_calls | CREATE TABLE `missed_calls` ( `id` int(10) unsigned NOT NULL auto_increment, `method` varchar(16) NOT NULL default '', `from_tag` varchar(64) NOT NULL default '', `to_tag` varchar(64) NOT NULL default '', `callid` varchar(128) NOT NULL default '', `sip_code` char(3) NOT NULL default '', `sip_reason` varchar(32) NOT NULL default '', `time` datetime NOT NULL default '0000-00-00 00:00:00', `src_ip` varchar(64) NOT NULL default '', `dst_user` varchar(64) NOT NULL default '', `dst_domain` varchar(128) NOT NULL default '', `src_user` varchar(64) NOT NULL default '', `src_domain` varchar(128) NOT NULL default '', `cdr_id` int(11) NOT NULL default '0', `dst_ouser` varchar(64) NOT NULL default '', PRIMARY KEY (`id`), KEY `mc_callid` (`callid`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 There are no errors in log, and nothing about acc module in debug mode. Cant get where is my mistake.

2 3

BUG: tls [tls_server.c:98]: Invalid connection state (bug in TCP code)
by MÉSZÁROS Mihály 12 Aug '09

12 Aug '09

Hello! Another problem I am using sip-router git version with the attached configuration files. I am getting the error mentioned in subject. Please see the debug log and my configuration files. Can anyone help me figure out what i a configured wrong? Misi Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 517 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <PUBLISH> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:alice@atlanta.test> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-260c1648321add52-1---d8754z->; state=6 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: ERROR: <script>: Misi: PUBLISH sip:alice@atlanta.test SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-260c1648321add52-1---d8754z-;rport#015#012Max-Forwards: 70#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: "alice"<sip:alice@atlanta.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=6470be16#015#012Call-ID: MTM5NDE0YjRjMmU2ZWNlOThjNDI1MGM2ZjAzMDBjZjA.#015#012CSeq: 16 PUBLISH#015#012Expires: 1200#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/pidf+xml#015#012Proxy-Authorization: Digest username="alice",realm="atlanta.test",nonce="4a81babd0000000bdc0c130f92356ce11fc7ea51a926c644",uri="sip:alice@atlanta.test",response="527ee7c53e6f621ef9ee161426da4afd",cnonce="9f418c76f9b91cd9c3d71fbb4d36f415",nc=00000002,qop=auth,algorithm=MD5#015#012SIP-If-Match: a.1250015022.397.2.8#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Event: presence#015#012Content-Length: 517#015#012#015#012<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:alice@atlanta.test'><tuple id='tae2c271a'><status><basic>open</basic></status></tuple><dm:person id='p7a66b539'><rpid:activities><rpid:busy/><rpid:on-the-phone/></rpid:activities><dm:note>On the phone</dm:note></dm:person></presence> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=9 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [33]; uri=[sip:alice@atlanta.test] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body ["alice"<sip:alice@atlanta.test>#015#012] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: siputils [checks.c:65]: no totag Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=3 global id=2 T start=0xffffffff Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field: cseq <CSeq>: <16> <PUBLISH> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: beginning of SIP-If-Match: yet=0x2d706973 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: middle of SIP-If-Match: yet=0x6d2d6669 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: end of SIP-If-Match Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_body : content_length=517 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:89]: found end of header Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=3518, isACK=0 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1149]: DEBUG: t_check_msg: msg id=3 global id=3 T end=(nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=6470be16 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [nonce.c:186]: comparing [4a81babd0000000bdc0c130f92356ce11fc7ea51a926c644] and [4a81babd0000000bdc0c130f92356ce11fc7ea51a926c644] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82bd628 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:68]: 3 columns returned from the query Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:153]: allocate 12 bytes for result names at 0x82f55d8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:163]: allocate 12 bytes for result types at 0x82f52fc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[0] at 0x82f4e4c Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4e4c)[0]=[password] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[1] at 0x82dbf88 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82dbf88)[1]=[email_address] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[2] at 0x82f5768 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f5768)[2]=[rpid] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82f4ebc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:119]: allocate 60 bytes for row values at 0x82f5390 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [bakker] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [sip:003631001001@atlanta.test] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:123]: HA1 string calculated: 67bb4b4b59a70b7f246682dfdd188df2 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:239]: our result = '527ee7c53e6f621ef9ee161426da4afd' Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:246]: authorization is OK Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:194]: nonce index= 11 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [index.c:187]: nonce already used Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:198]: nonce index not valid Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:79]: freeing 3 columns Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82f4e4c Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82dbf88 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[2] at 0x82f5768 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:92]: freeing result names at 0x82f55d8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:97]: freeing result types at 0x82f52fc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:52]: freeing 1 rows Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x82f5390 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:60]: freeing rows at 0x82f4ebc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:134]: freeing result set at 0x82bd628 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [index.c:128]: second= 16, sec_monit= 8, index= 12 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [challenge.c:107]: nonce index= 12 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [challenge.c:151]: 'Proxy-Authenticate: Digest realm="atlanta.test", nonce="4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2", qop="auth"#015#012' Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: sl [sl.c:314]: reply in stateless mode (sl) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 742 bytes written Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=742 fd=8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 407 Proxy Authentication Required#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-260c1648321add52-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: "alice"<sip:alice@atlanta.test>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.a62c#015#012From: "alice"<sip:alice@atlanta.test>;tag=6470be16#015#012Call-ID: MTM5NDE0YjRjMmU2ZWNlOThjNDI1MGM2ZjAzMDBjZjA.#015#012CSeq: 16 PUBLISH#015#012Proxy-Authenticate: Digest realm="atlanta.test", nonce="4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2", qop="auth"#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:alice@atlanta.test out_uri=sip:alice@atlanta.test via_cnt==1"#015#012#015#012 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 517 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <PUBLISH> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:alice@atlanta.test> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-8b6841568301eb55-1---d8754z->; state=6 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: ERROR: <script>: Misi: PUBLISH sip:alice@atlanta.test SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-8b6841568301eb55-1---d8754z-;rport#015#012Max-Forwards: 70#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: "alice"<sip:alice@atlanta.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=6470be16#015#012Call-ID: MTM5NDE0YjRjMmU2ZWNlOThjNDI1MGM2ZjAzMDBjZjA.#015#012CSeq: 17 PUBLISH#015#012Expires: 1200#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/pidf+xml#015#012Proxy-Authorization: Digest username="alice",realm="atlanta.test",nonce="4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2",uri="sip:alice@atlanta.test",response="cc8d63baac587662386d887e507bf60b",cnonce="1eeeeb8bf845982d90f185276c67bc78",nc=00000001,qop=auth,algorithm=MD5#015#012SIP-If-Match: a.1250015022.397.2.8#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Event: presence#015#012Content-Length: 517#015#012#015#012<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:alice@atlanta.test'><tuple id='tae2c271a'><status><basic>open</basic></status></tuple><dm:person id='p7a66b539'><rpid:activities><rpid:busy/><rpid:on-the-phone/></rpid:activities><dm:note>On the phone</dm:note></dm:person></presence> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=9 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [33]; uri=[sip:alice@atlanta.test] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body ["alice"<sip:alice@atlanta.test>#015#012] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: siputils [checks.c:65]: no totag Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=4 global id=3 T start=(nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field: cseq <CSeq>: <17> <PUBLISH> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: beginning of SIP-If-Match: yet=0x2d706973 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: middle of SIP-If-Match: yet=0x6d2d6669 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: end of SIP-If-Match Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_body : content_length=517 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:89]: found end of header Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=3519, isACK=0 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1149]: DEBUG: t_check_msg: msg id=4 global id=4 T end=(nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=6470be16 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [nonce.c:186]: comparing [4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2] and [4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82bd628 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:68]: 3 columns returned from the query Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:153]: allocate 12 bytes for result names at 0x82f55d8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:163]: allocate 12 bytes for result types at 0x82f52fc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[0] at 0x82f4ebc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4ebc)[0]=[password] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[1] at 0x82f5768 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f5768)[1]=[email_address] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[2] at 0x82dbf88 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82dbf88)[2]=[rpid] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82f4e4c Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:119]: allocate 60 bytes for row values at 0x82f5390 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [bakker] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [sip:003631001001@atlanta.test] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:123]: HA1 string calculated: 67bb4b4b59a70b7f246682dfdd188df2 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:239]: our result = 'cc8d63baac587662386d887e507bf60b' Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:246]: authorization is OK Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:194]: nonce index= 12 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:176]: set string AVP ""/13 = "sip:003631001001@atlanta.test" Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:79]: freeing 3 columns Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82f4ebc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82f5768 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[2] at 0x82dbf88 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:92]: freeing result names at 0x82f55d8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:97]: freeing result types at 0x82f52fc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:52]: freeing 1 rows Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x82f5390 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:60]: freeing rows at 0x82f4e4c Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:134]: freeing result set at 0x82bd628 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: uri_db [checks.c:137]: Digest username and URI username match Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: rr [loose.c:108]: No Route headers found Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: rr [loose.c:907]: There is no Route HF Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1383]: DEBUG: t_newtran: msg id=4 , global msg id=4 , T on entrance=(nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=3519, isACK=0 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_hooks.c:335]: DBG: trans=0xb585090c, callback type 1, id 0 entered Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [event_list.c:349]: start event= [presence/1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [publish.c:369]: SIP-If-Match header found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [publish.c:378]: existing etag = a.1250015022.397.2.8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [publish.c:390]: Expires header found, value= 1200 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82f56e4 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:68]: 2 columns returned from the query Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:153]: allocate 8 bytes for result names at 0x82f5768 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:163]: allocate 8 bytes for result types at 0x82f4ebc Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[0] at 0x82f4e84 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4e84)[0]=[body] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:121]: use DB1_BLOB result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[1] at 0x82f5730 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f5730)[1]=[sender] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82f4e10 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:119]: allocate 40 bytes for row values at 0x82f5960 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:168]: converting BLOB [<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:alice@atlanta.test'><tuple id='tae2c271a'><status><basic>open</basic></status></tuple><dm:person id='p7a66b539'><dm:note>Available</dm:note></dm:person></presence>] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:79]: freeing 2 columns Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82f4e84 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82f5730 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:92]: freeing result names at 0x82f5768 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:97]: freeing result types at 0x82f4ebc Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:52]: freeing 1 rows Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x82f5960 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:60]: freeing rows at 0x82f4e10 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:134]: freeing result set at 0x82f56e4 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [presentity.c:88]: etag= a.1250015022.390.2.9 / 20#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [presentity.c:102]: send 200OK reply Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [presentity.c:103]: etag= a.1250015022.390.2.9 - len= 20 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=4 global id=4 T start=0xb585090c Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1152]: DEBUG: t_check_msg: T already found! Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [mem/shm_mem.c:117]: WARNING:vqm_resize: resize(0) called Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:1355]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 643 bytes written Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=643 fd=8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 200 OK#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-8b6841568301eb55-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: "alice"<sip:alice@atlanta.test>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2965#015#012From: "alice"<sip:alice@atlanta.test>;tag=6470be16#015#012Call-ID: MTM5NDE0YjRjMmU2ZWNlOThjNDI1MGM2ZjAzMDBjZjA.#015#012CSeq: 17 PUBLISH#015#012Expires: 1200#015#012SIP-ETag: a.1250015022.390.2.9#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:alice@atlanta.test out_uri=sip:alice@atlanta.test via_cnt==1"#015#012#015#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:610]: DEBUG: reply sent out. buf=0x82f5c80: SIP/2.0 2..., shmem=0xb584f674: SIP/2.0 2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:620]: DEBUG: _reply_light: finished Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: sl [sl.c:309]: reply in stateful mode (tm) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [notify.c:1302]: found 0 dialogs( 0 in database and 0 in hash_table) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [notify.c:1321]: Could not find subs_dialog Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 544 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <INVITE> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:bob@biloxi.test> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-0b3389781d1b817e-1---d8754z->; state=6 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: ERROR: <script>: Misi: INVITE sip:bob@biloxi.test SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-0b3389781d1b817e-1---d8754z-;rport#015#012Max-Forwards: 70#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 1 INVITE#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/sdp#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Content-Length: 544#015#012#015#012v=0#015#012o=- 6 2 IN IP4 92.249.143.13#015#012s=CounterPath Bria Professional#015#012c=IN IP4 92.249.143.13#015#012t=0 0#015#012m=audio 4242 RTP/AVP 107 119 100 106 0 98 8 18 101#015#012a=alt:1 2 : bg/abBbq 9nirlFAG 192.168.1.7 4242#015#012a=alt:2 1 : wCTfkbfT 6n6JS41z 92.249.143.13 4242#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:119 BV32-FEC/16000#015#012a=rtpmap:100 SPEEX/16000#015#012a=rtpmap:106 SPEEX-FEC/16000#015#012a=rtpmap:98 iLBC/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=sendrecv#015#012a=x-rtp-session-id:C5EF9F1EE4C64633A625BEC3FF808160#015#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=9 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:bob@biloxi.test] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body [<sip:bob@biloxi.test>#015#012] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: siputils [checks.c:65]: no totag Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffff Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field: cseq <CSeq>: <1> <INVITE> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_body : content_length=544 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:89]: found end of header Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5d194242 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=47015, isACK=0 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1149]: DEBUG: t_check_msg: msg id=5 global id=5 T end=(nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:126]: credentials with given realm not found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [index.c:128]: second= 18, sec_monit= 8, index= 13 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [challenge.c:107]: nonce index= 13 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [challenge.c:151]: 'Proxy-Authenticate: Digest realm="atlanta.test", nonce="4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0", qop="auth"#015#012' Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: sl [sl.c:314]: reply in stateless mode (sl) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 724 bytes written Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=724 fd=8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 407 Proxy Authentication Required#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-0b3389781d1b817e-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: <sip:bob@biloxi.test>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d643#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 1 INVITE#015#012Proxy-Authenticate: Digest realm="atlanta.test", nonce="4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0", qop="auth"#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:bob@biloxi.test out_uri=sip:bob@biloxi.test via_cnt==1"#015#012#015#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 0 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <ACK> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:bob@biloxi.test> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-0b3389781d1b817e-1---d8754z->; state=6 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d643 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [65]; uri=[sip:bob@biloxi.test] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body [<sip:bob@biloxi.test>] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: sl [sl_funcs.c:327]: local ACK found -> dropping it! Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 544 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <INVITE> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:bob@biloxi.test> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-2a767f38e361686d-1---d8754z->; state=6 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: ERROR: <script>: Misi: INVITE sip:bob@biloxi.test SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport#015#012Max-Forwards: 70#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/sdp#015#012Proxy-Authorization: Digest username="alice",realm="atlanta.test",nonce="4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0",uri="sip:bob@biloxi.test",response="60ca1ab3ccd52ae04fded05e2056a49a",cnonce="41e9f5aabf671596242bb02343062a1b",nc=00000001,qop=auth,algorithm=MD5#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Content-Length: 544#015#012#015#012v=0#015#012o=- 6 2 IN IP4 92.249.143.13#015#012s=CounterPath Bria Professional#015#012c=IN IP4 92.249.143.13#015#012t=0 0#015#012m=audio 4242 RTP/AVP 107 119 100 106 0 98 8 18 101#015#012a=alt:1 2 : bg/abBbq 9nirlFAG 192.168.1.7 4242#015#012a=alt:2 1 : wCTfkbfT 6n6JS41z 92.249.143.13 4242#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:119 BV32-FEC/16000#015#012a=rtpmap:100 SPEEX/16000#015#012a=rtpmap:106 SPEEX-FEC/16000#015#012a=rtpmap:98 iLBC/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=sendrecv#015#012a=x-rtp-session-id:C5EF9F1EE4C64633A625BEC3FF808160#015#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=9 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:bob@biloxi.test] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body [<sip:bob@biloxi.test>#015#012] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: siputils [checks.c:65]: no totag Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=7 global id=5 T start=(nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field: cseq <CSeq>: <2> <INVITE> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_body : content_length=544 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:89]: found end of header Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5d194242 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=47012, isACK=0 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1149]: DEBUG: t_check_msg: msg id=7 global id=7 T end=(nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [nonce.c:186]: comparing [4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0] and [4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82bd628 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:68]: 3 columns returned from the query Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:153]: allocate 12 bytes for result names at 0x82f55d8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:163]: allocate 12 bytes for result types at 0x82f52b8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[0] at 0x82dbf88 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82dbf88)[0]=[password] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[1] at 0x82f4e4c Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4e4c)[1]=[email_address] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[2] at 0x82f4ebc Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4ebc)[2]=[rpid] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82f5768 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:119]: allocate 60 bytes for row values at 0x82f5960 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [bakker] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [sip:003631001001@atlanta.test] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:123]: HA1 string calculated: 67bb4b4b59a70b7f246682dfdd188df2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:239]: our result = '60ca1ab3ccd52ae04fded05e2056a49a' Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:246]: authorization is OK Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:194]: nonce index= 13 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:176]: set string AVP ""/13 = "sip:003631001001@atlanta.test" Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:79]: freeing 3 columns Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82dbf88 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82f4e4c Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[2] at 0x82f4ebc Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:92]: freeing result names at 0x82f55d8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:97]: freeing result types at 0x82f52b8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:52]: freeing 1 rows Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x82f5960 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:60]: freeing rows at 0x82f5768 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:134]: freeing result set at 0x82bd628 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: uri_db [checks.c:137]: Digest username and URI username match Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: rr [loose.c:108]: No Route headers found Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: rr [loose.c:907]: There is no Route HF Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==9 && [biloxi.test] == [127.0.0.1] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==14 && [biloxi.test] == [195.111.158.41] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==9 && [biloxi.test] == [127.0.0.1] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==14 && [biloxi.test] == [195.111.158.41] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==9 && [biloxi.test] == [127.0.0.1] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==14 && [biloxi.test] == [195.111.158.41] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==9 && [biloxi.test] == [127.0.0.1] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==14 && [biloxi.test] == [195.111.158.41] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [forward.c:318]: check_self: host != me Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1383]: DEBUG: t_newtran: msg id=7 , global msg id=7 , T on entrance=(nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=47012, isACK=0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_hooks.c:335]: DBG: trans=0xb584cf18, callback type 1, id 0 entered Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_funcs.c:348]: SER: new INVITE Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [mem/shm_mem.c:117]: WARNING:vqm_resize: resize(0) called Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 572 bytes written Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=572 fd=8 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 100 trying -- your call is important to us#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:bob@biloxi.test out_uri=sip:bob@biloxi.test via_cnt==1"#015#012#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:610]: DEBUG: reply sent out. buf=0x82f5c80: SIP/2.0 1..., shmem=0xb58505ec: SIP/2.0 1 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:620]: DEBUG: _reply_light: finished Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:560]: dns_hash_find(biloxi.test(11), 35), h=599 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2674]: naptr_iterate: found a valid sip NAPTR rr _sip._tcp.biloxi.test, proto 2 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1091]: naptr_choose(o: -1 w: -1 p:0 , o: 20 w:50 p:2) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1107]: naptr_choose: changed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2674]: naptr_iterate: found a valid sip NAPTR rr _sip._udp.biloxi.test, proto 1 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1091]: naptr_choose(o: 20 w: 50 p:2 , o: 30 w:50 p:1) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1102]: naptr_choose: no change Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2674]: naptr_iterate: found a valid sip NAPTR rr _sips._tcp.biloxi.test, proto 3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1091]: naptr_choose(o: 20 w: 50 p:2 , o: 10 w:50 p:3) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1107]: naptr_choose: changed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2688]: naptr_iterate: choosed NAPTR rr _sips._tcp.biloxi.test, proto 3 tried: 0x0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:560]: dns_hash_find(_sips._tcp.biloxi.test(22), 33), h=208 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2246]: dns_srv_get_nxt_rr(0xb5864a2c, 0, 0, 559080590): selected 0/1 in grp. 0 (rand_w=0, rr=0xb5864a70 p=10 w=0 rsum=0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:560]: dns_hash_find(b1.biloxi.test(14), 1), h=386 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2861]: dns_a_resovle(b1.biloxi.test, 0) returning 0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3115]: dns_srv_resolve_ip("_sips._tcp.biloxi.test", 0, 0), ret=0, ip=195.111.158.43 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3350]: dns_naptr_sip_resolve(biloxi.test, 0, 0), srv0, ret=0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: <script>: new branch at sip:bob@biloxi.test Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:2379]: create_via_hf: id added: <;i=2>, rcv proto=3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1778]: tcp_send: no open tcp connection found, opening new one Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [ip_addr.c:116]: tcpconn_new: new tcp connection: 195.111.158.43 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1050]: tcpconn_new: on port 5061, type 3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:3014]: handle_ser_child: read response= b5865b20, 2, fd 68 from 36 (390) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: BUG: tls [tls_server.c:98]: Invalid connection state (bug in TCP code) Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:1349]: tcpconn_add: hashes: 1306:736:1196, 9 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tls [tls_server.c:145]: Delayed init failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [io_wait.h:361]: DBG: io_watch_add(0x8228560, 68, 2, 0xb5865b20), fd_no=55 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb5865b20 n=-1 fd=13 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012INVITE sip:bob@biloxi.test SIP/2.0#015#012Record-Route: <sip:195.111.158.41:5061;transport=tls;lr=on;ftag=5d194242>#015#012Via: SIP/2.0/TLS 195.111.158.41:5061;branch=z9hG4bK4a7b.1036c875.0;i=2#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;received=92.249.143.13;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783#015#012Max-Forwards: 69#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/sdp#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Content-Length: 544#015#012P-hint: outbound#015#012#015#012v=0#015#012o=- 6 2 IN IP4 92.249.143.13#015#012s=CounterPath Bria Professional#015#012c=IN IP4 92.249.143.13#015#012t=0 0#015#012m=audio 4242 RTP/AVP 107 119 100 106 0 98 8 18 101#015#012a=alt:1 2 : bg/abBbq 9nirlFAG 192.168.1.7 4242#015#012a=alt:2 1 : wCTfkbfT 6n6JS41z 92.249.143.13 4242#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:119 BV32-FEC/16000#015#012a=rtpmap:100 SPEEX/16000#015#012a=rtpmap:106 SPEEX-FEC/16000#015#012a=rtpmap:98 iLBC/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=sendrecv#015#012a=x-rtp-session-id:C5EF9F1EE4C64633A625BEC3FF808160#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: <core> [tcp_main.c:2166]: ERROR: tcp_send: failed to send on 0xb5865b20 (195.111.158.41:59267->195.111.158.43:5061): Operation now in progress (115) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tm [../../forward.h:163]: msg_send: ERROR: tcp_send failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:3014]: handle_ser_child: read response= b5865b20, -2, fd -1 from 36 (390) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_fwd.c:961]: t_send_branch: send to 195.111.158.43:5061 (3) failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: ERROR: <core> [tcp_main.c:3027]: handle_ser_child: ERROR: received CON_ERROR for 0xb5865b20 (id 9), refcnt 2 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2861]: dns_a_resovle(b1.biloxi.test, 1) returning -1 Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8228560, 68, -1, 0x10) fd_no=56 called Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2246]: dns_srv_get_nxt_rr(0xb5864a2c, 0, 0, 559080593): selected 0/1 in grp. 1 (rand_w=0, rr=0xb5864a98 p=20 w=0 rsum=0) Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: tls [tls_server.c:689]: Closing SSL connection Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:560]: dns_hash_find(b2.biloxi.test(14), 1), h=281 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2861]: dns_a_resovle(b2.biloxi.test, 0) returning 0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3115]: dns_srv_resolve_ip("_sips._tcp.biloxi.test", 1, 0), ret=0, ip=195.111.158.44 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3256]: dns_sip_resolve(biloxi.test, 1, 0), srv, ret=0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:2379]: create_via_hf: id added: <;i=2>, rcv proto=3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_fwd.c:974]: t_send_branch: send on branch 0 failed, adding another branch with another ip Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1778]: tcp_send: no open tcp connection found, opening new one Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [ip_addr.c:116]: tcpconn_new: new tcp connection: 195.111.158.44 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1050]: tcpconn_new: on port 5061, type 3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[390]: BUG: tls [tls_server.c:98]: Invalid connection state (bug in TCP code) Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:3014]: handle_ser_child: read response= b5865b20, 2, fd 68 from 36 (390) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tls [tls_server.c:145]: Delayed init failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:1349]: tcpconn_add: hashes: 1693:359:2697, 10 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb5865b20 n=-1 fd=13 Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [io_wait.h:361]: DBG: io_watch_add(0x8228560, 68, 2, 0xb5865b20), fd_no=55 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012INVITE sip:bob@biloxi.test SIP/2.0#015#012Record-Route: <sip:195.111.158.41:5061;transport=tls;lr=on;ftag=5d194242>#015#012Via: SIP/2.0/TLS 195.111.158.41:5061;branch=z9hG4bK4a7b.1036c875.1;i=2#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;received=92.249.143.13;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783#015#012Max-Forwards: 69#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/sdp#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Content-Length: 544#015#012P-hint: outbound#015#012#015#012v=0#015#012o=- 6 2 IN IP4 92.249.143.13#015#012s=CounterPath Bria Professional#015#012c=IN IP4 92.249.143.13#015#012t=0 0#015#012m=audio 4242 RTP/AVP 107 119 100 106 0 98 8 18 101#015#012a=alt:1 2 : bg/abBbq 9nirlFAG 192.168.1.7 4242#015#012a=alt:2 1 : wCTfkbfT 6n6JS41z 92.249.143.13 4242#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:119 BV32-FEC/16000#015#012a=rtpmap:100 SPEEX/16000#015#012a=rtpmap:106 SPEEX-FEC/16000#015#012a=rtpmap:98 iLBC/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=sendrecv#015#012a=x-rtp-session-id:C5EF9F1EE4C64633A625BEC3FF808160#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: <core> [tcp_main.c:2166]: ERROR: tcp_send: failed to send on 0xb5865b20 (195.111.158.41:55898->195.111.158.44:5061): Operation now in progress (115) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tm [../../forward.h:163]: msg_send: ERROR: tcp_send failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:3014]: handle_ser_child: read response= b5865b20, -2, fd -1 from 36 (390) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_fwd.c:961]: t_send_branch: send to 195.111.158.44:5061 (3) failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: ERROR: <core> [tcp_main.c:3027]: handle_ser_child: ERROR: received CON_ERROR for 0xb5865b20 (id 10), refcnt 2 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2861]: dns_a_resovle(b2.biloxi.test, 1) returning -1 Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8228560, 68, -1, 0x10) fd_no=56 called Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3115]: dns_srv_resolve_ip("_sips._tcp.biloxi.test", 2, 1), ret=-1, ip= Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: tls [tls_server.c:689]: Closing SSL connection Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3256]: dns_sip_resolve(biloxi.test, 2, 1), srv, ret=-1 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tm [t_fwd.c:980]: ERROR: t_send_branch: sending request on branch 1 failed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_funcs.c:358]: ERROR:tm:t_relay_to: t_forward_nonack returned error Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_funcs.c:366]: -477 error reply generation delayed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: sl [sl_funcs.c:283]: error text is Unfortunately error on sending to next hop occurred (477/SL) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 636 bytes written Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=636 fd=8 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/SL)#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: <sip:bob@biloxi.test>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.573f#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:bob@biloxi.test out_uri=sip:bob@biloxi.test via_cnt==1"#015#012#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1536]: t_unref: delayed error reply generation(-477) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:1355]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 636 bytes written Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=636 fd=8 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM)#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: <sip:bob@biloxi.test>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-573f#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:bob@biloxi.test out_uri=sip:bob@biloxi.test via_cnt==1"#015#012#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_hooks.c:252]: DBG: trans=0xb584cf18, callback type 128, id 0 entered Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:610]: DEBUG: reply sent out. buf=0x82f6088: SIP/2.0 4..., shmem=0xb58505ec: SIP/2.0 4 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:620]: DEBUG: _reply_light: finished Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <ACK> Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:bob@biloxi.test> Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-2a767f38e361686d-1---d8754z->; state=6 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=f8f2ab2c1295e90ed7dbb499b30f44b2.573f Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [65]; uri=[sip:bob@biloxi.test] Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body [<sip:bob@biloxi.test>] Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: sl [sl_funcs.c:327]: local ACK found -> dropping it! Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 447 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <PUBLISH> Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:alice@atlanta.test> Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-0d1d4d59c544e42e-1---d8754z->; state=6 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via # # $Id$ # # This a very basic config file w aliases and anamed route but # w/o authentication, accounting, database, multi-domain support etc. # Please refer to sip-router.cfg for a more complete example # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) memdbg=6 # memory debug message level memlog=6 # memory statistics log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 user=ser group=ser #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #TLS enable_tls=yes #TCP settings disable_tcp=no tcp_async=no tcp_send_timeout=2 #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) tcp_delayed_ack=no tcp_fd_cache=no # Alias alias=atlanta.test:5061 alias=atlanta.test:5060 #Tells how long to keep negative DNS responses in cache. If set to 0, disables caching of negative responses. #Default is 60 (seconds). dns_cache_negative_ttl=5 #Number of dns retransmissions before giving up. dns_retr_no=3 #Time in seconds before retrying a dns request. dns_retr_time=3 #dns_srv_loadbalancing Enable dns srv weight based load balancing dns_srv_lb = yes #If it is set to 'yes' and a DNS lookup fails, it will retry it for ipv6 (AAAA record) dns_try_ipv6=yes #Enable NAPTR support according to RFC 3263 dns_try_naptr = yes #Set preference for each protocol when doing naptr lookups. #By default dns_udp_pref=30, dns_tcp_pref=20, dns_tls_pref=10 and dns_sctp_pref=20. #To completely ignore NAPTR records for a specific protocol, set the corresponding protocol preference to -1 (or any other negative number). dns_udp_pref = 1 dns_tcp_pref = 1 dns_tls_pref = 1 dns_sctp_pref = 1 #If set to 'no', the search list in '/etc/resolv.conf' will be ignored. #(-> fewer lookups -> gives up faster). Default value is 'yes'. dns_use_search_list=no #default off use_dns_failover=yes # ------------------ module loading ---------------------------------- #loadpath "modules:modules_s" #loadpath "/usr/lib/ser/modules:/usr/lib/ser/modules_s" # common modules loadpath "/usr/lib/ser/modules" loadmodule "tm" loadmodule "utils" loadmodule "db_mysql" loadmodule "avpops.so" loadmodule "tls" # Kamailio modules loadpath "/usr/lib/ser/modules_k" loadmodule "sl" loadmodule "rr" loadmodule "usrloc" loadmodule "registrar" loadmodule "maxfwd" loadmodule "textops" loadmodule "uri_db.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "pv.so" loadmodule "siputils.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_mwi.so" loadmodule "presence_dialoginfo.so" loadmodule "xlog.so" # ser modules loadpath "/usr/lib/ser/modules_s" loadmodule "ctl" loadmodule "cfg_rpc" loadmodule "auth_identity" #loadmodule "sl" #loadmodule "sanity" #loadmodule "registrar" # ----------------- setting module-specific parameters --------------- # -- tls params -- modparam("tls","config","/etc/ser/tls.cfg") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ctl params # by default ctl listens on unixs:/tmp/sip-router_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/sip-router_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/sip-router_fifo") # listen on tcp, localhost #modparam("ctl", "binrpc", "tcp:localhost:2046") # ----- registrar params ----- #modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- usrloc params ----- #modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- auth params ----- modparam("auth", "rpid_avp", "$avp(i:13)") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") modparam("auth_db", "load_credentials", "$avp(i:13)=rpid;email_address") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ modparam("alias_db", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence", "db_url","mysql://opensips:opensipsrw@localhost/opensips") modparam("presence", "server_address", "sip:atlanta.test") modparam("presence_xml", "db_url","mysql://opensips:opensipsrw@localhost/opensips") modparam("presence_xml", "force_active", 1) # ----- auth identity params ----- modparam("auth_identity","cainfo_path","/etc/ssl/certs/ca-certificates.crt") #TODO web server publicating CERTIFICATE. modparam("auth_identity","certificate_url","https://www.atlanata.test/mycert.der") modparam("auth_identity","certificate_path","/etc/ser/tls/cert.pem") modparam("auth_identity","privatekey_path","/etc/ser/tls/privkey.pem") ####### Routing Logic ######## # main request routing logic route{ xlog("L_ERR", "Misi: $mb\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (msg:len >= 4096 ) { sl_send_reply("513", "Message too big"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } else if (is_method("INVITE")) { # even if in most of the cases is useless, do RR for # re-INVITEs alos, as some buggy clients do change route set # during the dialog. record_route(); } # route it out to whatever destination was set by loose_route() # in $du (destination URI). route(1); } else { /* uncomment the following lines if you want to enable presence */ if (is_method("SUBSCRIBE") && $rd == "atlanta.test") { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after # a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction -> # ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authenticate if from local subscriber (uncomment to enable auth) # authenticate all initial non-REGISTER request that pretend to be # generated by local subscriber (domain from FROM URI is local) if (!(method=="REGISTER") && from_uri==myself) /*no multidomain version*/ ##if (!(method=="REGISTER") && is_from_local()) /*multidomain version*/ { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } # preloaded route checking if (loose_route()) { xlog("L_ERR", "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]"); if (!is_method("ACK")) sl_send_reply("403","Preload Route denied"); exit; } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) ## replace with following line if multi-domain support is used ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain ## uncomment this if you want to enable presence server ## and comment the next 'if' block ## NOTE: uncomment also the definition of route[2] from below if( is_method("PUBLISH|SUBSCRIBE")) route(2); #if (is_method("PUBLISH")) #{ # sl_send_reply("503", "Service Unavailable"); # exit; #} if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if (!check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if (!save("location")) sl_reply_error(); exit; } if ($rU=="") { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) alias_db_lookup("dbaliases"); if(uri=~"sip:0.*@.*"){ sethost("195.111.158.3"); strip(1); # Append Remote-Party-ID header field append_rpid_hf("<", ">;party=calling;id-type=subscriber;screen=yes;privacy=off"); /* Remote-Party-ID: <sip:003646565111@195.111.104.13>;party=calling;screen=no;privacy=off */ route(1); exit; } if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } # Presence route /* uncomment the whole following route for enabling presence NOTE: do not forget to enable the call of this route from the main route */ route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); # Removed command. Not Supported anymore. # At the end of script the transaction is automaticaly relased by TM. #t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); # Removed command. Not Supported anymore. # At the end of script the transaction is automaticaly relased by TM. #t_release(); } exit; } branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("L_ERR", "Reply Misi: $mb\n"); xlog("incoming reply\n"); } failure_route[1] { if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## # do not set the missed call flag again ## t_relay(); ##} } ####################################Ser########################################## # ## ------------------------- request routing logic ------------------- # ## main routing logic # #route{ # # # initial sanity checks -- messages with # # max_forwards==0, or excessively long requests # if (!mf_process_maxfwd_header("10")) { # sl_send_reply("483","Too Many Hops"); # break; # } # if (msg:len >= 4096 ) { # sl_send_reply("513", "Message too big"); # break; # } # # # we record-route all messages -- to make sure that # # subsequent messages will go through our proxy; that's # # particularly good if upstream and downstream entities # # use different transport protocol # if (!method=="REGISTER") record_route(); # # # subsequent messages withing a dialog should take the # # path determined by record-routing # if (loose_route()) { # # mark routing logic in request # append_hf("P-hint: rr-enforced\r\n"); # route(FORWARD); # break; # } # # if (!uri==myself) { # # mark routing logic in request # append_hf("P-hint: outbound\r\n"); # route(FORWARD); # break; # } # # # if the request is for other domain use UsrLoc # # (in case, it does not work, use the following command # # with proper names and addresses in it) # if (uri==myself) { # # if (method=="REGISTER") { # save_contacts("location"); ## break; # } # # # native SIP destinations are handled using our USRLOC DB # if (!lookup_contacts("location")) { # sl_send_reply("404", "Not Found"); # break; # } ## append_hf("P-hint: usrloc applied\r\n"); # } # route(FORWARD); #} # #route[FORWARD] #{ # # send it out now; use stateful forwarding as it works reliably # # even for UDP2TCP # if (!t_relay()) { # sl_reply_error(); ## } #} # # $Id$ # # Example SER TLS Configuration File # # This is the default server domain, settings # in this domain will be used for all incoming # connections that do not match any other server # domain in this configuration file. # # We do not enable anything else than TLSv1 # over the public internet. Clients do not have # to present client certificates by default. # [server:default] method = TLSv1 verify_certificate = yes require_certificate = no private_key = /etc/ser/tls/privkey.pem certificate = /etc/ser/tls/cert.pem ca_list = /etc/ser/tls/calist.pem # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] verify_certificate = yes require_certificate = yes # This is example server domain for TLS connections # received from the loopback interface. We allow # the use of SSLv2 and SSLv3 protocols here, we do # not require that clients present client certificates # but if they present it it must be valid. We also use # a special certificate and CA list for loopback # interface. # #[server:127.0.0.1:5061] #method = SSLv23 #verify_certificate = yes #require_certificate = no #private_key = local_key.pem #certificate = local_cert.pem #verify_depth = 3 #ca_list = local_ca.pem # Special settings for the iptel.org public SIP # server. We do not verify the certificate of the # server because it can be expired. The server # implements authentication using SSL client # certificates so configure the client certificate # that was given to use by iptel.org staff here. # #[client:195.37.77.101:5061] #verify_certificate = no #certificate = iptel_client.pem #private_key = iptel_key.pem #ca_list = iptel_ca.pem

2 3

sip-router select
by MÉSZÁROS Mihály 12 Aug '09

12 Aug '09

Hi! Can any of you explain how select framework is working? Where can i find any documentation about it. I am experiencing problem when i am try using it. Any help highly appreciated! Regards, Misi

2 4

[Kamailio-Users] HOWTO rewrite to: header
by alexander merkulov 12 Aug '09

12 Aug '09

here is correct reqwest and reqwest via kamailio is there any way to rewrite To: field in kamailio because now it nto work correct reqwest(asterisk) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a From: "Unknown" <sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com> >;tag=as366a0ba0 To: <sip:34249@193.110.78.12:8484> Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 incorrect reqwest(multi-homed kamailio, asterisk originate to internal eth) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0 From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as6075df60 Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> CSeq: 102 CANCEL Max-Forwards: 70 asterisk Content-Length: 0 -- Merkulov Alexander

6 21

[Kamailio-Users] kamailio auth with service provider
by Lance Evans 12 Aug '09

12 Aug '09

Hey Guys, Apparently Kamailio cannot authenticate with my service provider. Can this not be done with a sip trunk or peering!? if so how do i setup a trunk with kamailio!? and how do i setup peering?! Thanks

2 1

[Kamailio-Users] Fwd: Getting Error 477 while registering Softphones
by Daniel-Constantin Mierla 12 Aug '09

12 Aug '09

... fwd to list ... reply all button mismatch ... ---------- Forwarded message ---------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Wed, Aug 12, 2009 at 8:27 AM Subject: Re: [Kamailio-Users] Getting Error 477 while registering Softphones To: Meftah Tayeb <tayeb.meftah(a)gmail.com> Hello, add: alias=tayeb.blogdns.com in your config file or use the domain module: http://kamailio.org/docs/modules/1.5.x/domain.html http://kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#alias Cheers, Daniel On Wed, Aug 12, 2009 at 1:15 AM, Meftah Tayeb <tayeb.meftah(a)gmail.com>wrote: > hello, > i'm getting Error 477 while registering a Softphone from outside my domain > i added my domain, tayeb.blogdns.com to the domain name variable, > sip_domain in kamctlrc.conf > created the DB, started, and added user 1000 and 1001 > in registration: > for internal use, i use 192.168.1.3 (Kamailio IP Address) > and for external use i use my domain > for internal, is working corectly but while i register a external softphone > is getting Error 477 > what is this error message? > how i can fix it? > please try to help me, any help is welcome > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4327 (20090811) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Getting Error 477 while registering Softphones
by Meftah Tayeb 11 Aug '09

11 Aug '09

hello, i'm getting Error 477 while registering a Softphone from outside my domain i added my domain, tayeb.blogdns.com to the domain name variable, sip_domain in kamctlrc.conf created the DB, started, and added user 1000 and 1001 in registration: for internal use, i use 192.168.1.3 (Kamailio IP Address) and for external use i use my domain for internal, is working corectly but while i register a external softphone is getting Error 477 what is this error message? how i can fix it? please try to help me, any help is welcome __________ Information from ESET NOD32 Antivirus, version of virus signature database 4327 (20090811) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

1 0

[Kamailio-Users] dialog info + Grandstreams = freezing
by David 11 Aug '09

11 Aug '09

Hello, I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS 1.5.2-tls compilde on Ubuntu 8.04 I am trying to setup presence_dialoginfo with my Grandstreams, Snom and Linksys. I have a 4 phones on the server. 101 - Linksys SPA962 ( 6.1.5a ) 102 - Grandstream GXP2020 ( 1.2.1.4 ) 103 - Grandstream GXP2000 ( 1.2.1.4 ) 104 - Grandstream GXP2000 ( 1.1.6.46 ) 105 - Snom 360 ( 7.3.23 ) My Kamailio deals with registrations, NAT and BLF everything else is sent to one of two asterisk boxes. I use the dispatcher module for this. This means that when I call one extension to the other, both call legs from asterisk are going through Kamailio as separate calls. But to divide my customers, the usernames are different from the URL that the user types. For example the customer dials '101' but it is changed to testspace.101 when it comes back from asterisk. So Kamailio would have two calls in the event that 101 dials 102. sip:testspace.101@myserver to 102 ( this is sent to asterisk ) sip:testspace.101@myserver to testspace.102 ( this is coming back from asterisk ) Something is horribly wrong. I have the following problems : 1. If 102 calls 103, when 103 answers both phones hang for about 2 minutes 2. If 105 calls 101, 101 BLF comes back to the inactive state ( green on the Linksys and dark on the Snom), but the orange light stays on on the Snom and it thinks the call is still active ( the light is on, but the call is over ) 3. If any extension calls any extension and I try a call pickup, it fails. It looks like the Linksys is sending a NOTIFY to pickup the call ( I thought it was supposed to send an invite... ? ) Looking at the logs it looks like Kamailio is sending out so many NOTIFYs that it is crashing the Grandstreams, and causing the Snom to act funny. Here are some experts from my config file : root@kamailio-dev:/etc/kamailio# grep dialog * kamailio.cfg:# * avp value for dialogs is still not correct kamailio.cfg:loadmodule "dialog.so" kamailio.cfg:loadmodule "presence_dialoginfo.so" kamailio.cfg:loadmodule "pua_dialoginfo.so" kamailio.cfg:#modparam("pua_dialoginfo", "include_localremote", 0) kamailio.cfg:#modparam("pua_dialoginfo", "include_tags", 0) kamailio.cfg:#modparam("pua_dialoginfo", "include_callid", 0) kamailio.cfg:modparam("dialog", "dlg_flag", 4) kamailio.cfg:modparam("dialog", "db_mode", 1) kamailio.cfg:modparam("dialog", "timeout_avp", "$avp(i:10)") # I still haven't figured out how to set $avp(i:10) kamailio.cfg:modparam("pua_dialoginfo", "override_lifetime", 300) kamailio.cfg:modparam("presence_dialoginfo", "force_single_dialog", 1) kamailio.cfg:modparam("pua_dialoginfo", "caller_confirmed", 1) kamailio.cfg:modparam("auth_db|usrloc|acc|domain|avpops|presence|presence_xml|pua|dialog", "db_url", kamailio.cfg:# Flag 4 = Mark the current request for a dialog kamailio.cfg: # sequential request withing a dialog should the set flag looks like this : if ( ds_is_from_list() ) { xlog("L_INFO", "Coming from asterisk"); if ( is_method("INVITE")) { setflag(4); } } So the dialog flag is only set for the leg coming back from asterisk. When a notify comes in : if(is_method("NOTIFY") ) { if (! t_newtran()) { sl_reply_error(); exit; }; t_reply("200", "OK"); t_release(); exit ; } Publish and subscribe are like this : if( is_method("PUBLISH") || is_method("SUBSCRIBE") ) { route(5); exit; } route[5] { # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Ignoring PUBLISH/SUBSCRIBE on retransmition - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; }; append_to_reply("Contact: <sip:myserver.tld:5060>\r\n"); if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } else { } exit; } I also have NAT checking for those telephones where stun isn't enough. Before I reach publish/subscribe/invite/notify, I also call setbflag() and sometimes call fix_nated_contact(). Additionnally, I have a block if code before my presence stuff if ( has_totag() && loose_route()) { t_relay(); }. If sip.conf:canreinvite=yes, the grandstreams freeze so long that the server times out, and the BLFs get really messed up. if sip:canreinvite=no the grandstreams only freeze for about 30 seconds. Obviously I am doing something wrong, but despite having searched google for endless hours, and poured over documentation, I can not seem to find what I did wrong. I would really appreciate if someone could shed light on my problem. I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS 1.5.2-tls compilde on Ubuntu 8.04 Thanks, David

2 3

[Kamailio-Users] problem regarding openser+sip+raduis project
by Sanyog Kale 10 Aug '09

10 Aug '09

sir i am working on project Build SIP-based VOIP Service With RADIUS AAA Using Kamailio (OpenSER) And FreeRadius as per tutorial given by you. but the problem is 1.i am not understanding how to intialize modules for freeradius 2.while testing radius server where to make file named digest 3.the openser configuration which is given by you should be stored in which file? 4.and after all configurations done how to test all the parameters configured hoping for reply regards sanyog

4 3

[Kamailio-Users] Traffic parsing
by michel freiha 09 Aug '09

09 Aug '09

Hi all, I would like to ask you please how to configure kamailio in order to send signaling traffic to a specific gateway and rtp traffic to another gateway? Kindly note that I'm using media proxy for rtp and carrierroute for sending traffic to different gateways Regards

4 5

[Kamailio-Users] Siremis accounting
by Lance Evans 07 Aug '09

07 Aug '09

Hey guys Nothing seems to be showing in the Accounting tabs (in any of the subtabs) Any ideas!?

3 2

[Kamailio-Users] Siremis not displaying anything in Online SIP Users
by Lance Evans 07 Aug '09

07 Aug '09

Hey Guys, Hope someone has time for a real newbie to VOIP, kamailio and siremis!? I got Kamailio and Siremis work ok. Fighting with Siremis accounting still... but for i cant seem see anything in the Online SIP Users when i'm making calls etc. ubuntu server 8.10 mysql v5.0.67 php v5.2.6-2 Kamailio 1.5.2 notls (built from source) + siremis 0.9.3 let me know if you guys need anything else!? Thanks

3 2

sip-router + tls
by MÉSZÁROS Mihály 07 Aug '09

07 Aug '09

Hi All! I have on question regarding the TLS module. loadpath "/usr/lib/ser/modules" loadmodule "avpops.so" loadmodule "tls" _but_ i can use only tlsops setting parameters. modparam("tlsops","config","/etc/ser/tls.cfg") tls module name in tlsops? modules/tls_mod.c /* * Module interface */ struct module_exports exports = { "tlsops", Is this a typo or some work in progress? Is it a bug, or a feature? Thanks, Misi

3 4

sip-router + tls
by MÉSZÁROS Mihály 07 Aug '09

07 Aug '09

Hi All! I have on question regarding the TLS module. loadpath "/usr/lib/ser/modules" loadmodule "avpops.so" loadmodule "tls" _but_ i can use only tlsops setting parameters. modparam("tlsops","config","/etc/ser/tls.cfg") tls module name in tlsops? modules/tls_mod.c /* * Module interface */ struct module_exports exports = { "tlsops", Is this a typo or some work in progress? Is it a bug, or a feature? Thanks, Misi

1 0

[Kamailio-Users] Local routing
by Martin Werne 06 Aug '09

06 Aug '09

Hi! I'm planning a new platform based on Kamailio, to replace an existing Cisco-platform. - A total of appr. 2000 ATAs/IP Phones (today appr. 300). - Routing to various providers/gateways via custom made OSP-server. - Two Kamailio-servers on different geographic locations. - Load-balancing/Failover of ATA's by use of DNS-SRV. The main "problem" I see, is how local calls between ATA's registered on each of the Kamailio-servers is routed. How can Kamailio1 route a call to an ATA registered on Kamailio2? What is best in this scenario? 1. If user not found on local Kamailio-server, forward call to next Kamailio-server. 2. Let the Kamailio-servers share the same database (same location-table)? 3. Let the Kamailio-servers copy the REGISTER to the other Kamailio-server(s) so that the other Kamailio-server(s) update their own location-table. I would prefer not to use alternative 1... If alternative 3 is the best, is 't_replicate' the way to do it? What if I add a third Kamailio-server, will that change the recommendation? Martin

4 4

[Kamailio-Users] cfgutils module
by karhu 06 Aug '09

06 Aug '09

Hello, I'm trying something but doesn't work :s Here is my idea : In my cfg file i added the following lines : if (is_gflag("10")) xlog("L_INFO", "gflag is 10"); I'm using MI Commands to set gflag to 10 so : $ kamctl fifo set_gflag 10 After that if i look on my log file there is no "gflag is 10" !! Am i using the module and MI commands in the right way ? Thanks -- View this message in context: http://www.nabble.com/cfgutils-module-tp24823254p24823254.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 11

nat_uac_test mode parameter 16
by Andrew O. Zhukov 05 Aug '09

05 Aug '09

I just spend a lot of time to fix a trouble with Cisco AS5300 (AS5350) BYE message disapearing. The same not resolved case here: http://lists.iptel.org/pipermail/serusers/2005-July/021698.html The source of trouble was in the nat_uac_test parameter 16: Here is the part of cisco log: Jul 31 12:16:00.726 UKR: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 193.28.184.13:5060, local_port 50058 --Skipped--- Jul 31 12:16:00.738 UKR: Sent: INVITE sip:3039100@193.28.184.13:5060 SIP/2.0 Via: SIP/2.0/UDP 193.28.184.21:5060 From: <sip:2308228@193.28.184.21>;tag=12E601F8-10C7 To: <sip:3039100@193.28.184.13> Via header the same in the all availible IOS-es. As the result openser considerate the Cisco is out of the nat box. The question is: Can someone show me the case when this parameter is necessary, before I'll switch it of on the production server? Thanks in advance. Andrew O. Zhukov.

3 6

[Kamailio-Users] $hdr error if used with an avp
by catalina oancea 04 Aug '09

04 Aug '09

Hello I need to get the value of a header using a dynamic variable. I have the name of the header in an avp and I get the value using $hdr($avp(s:headername))). This worked in kamailio 1.4.0, I am sure of it because I tried it now on 1.4.0. But now in kamailio 1.5.1 I get an error when trying to retrieve the value: /usr/sbin/kamailio[32761]: ERROR:core:pv_get_spec_name: name type is PV_NAME_OTHER - cannot resolve /usr/sbin/kamailio[32761]: ERROR:pv:pv_get_hdr: invalid name Does anybody have any idea what the problem might be? Thanks

2 4

[Kamailio-Users] QjSimple 0.6.3 - sip phone for linux, macosx and windows
by Daniel-Constantin Mierla 03 Aug '09

03 Aug '09

Thanks to Klaus Darilion, QjSimple 0.6.3 works on most of desktop OS: Linux(es), Windows and Mac OS X. It is a very handy application, especially for developers, but not olny, with simple UI and debugger window. A perfect replacement (for my very old good Linux testing tool) KPhone since QjSimple has SIMPLE presence support, tls ... more at: http://www.ipcom.at/index.php?id=560/ Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/

3 6

[Kamailio-Users] dokuwiki, bug tracker and git
by Daniel-Constantin Mierla 03 Aug '09

03 Aug '09

Hello, I installed two plugins for dokuwiki and flyspray bug tracker to reference easier the git repository and bugs. For FlySpray, when you describe or comment a bug you can reference to a git commit by using: GIT#hash - hash must be at list the first six characters of git commit hash value Unfortunately FlySpray accepts only plain text in the "Close Task" description, so the git reference does not work there, yet. In dokuwiki same syntax can be used. In addition, you can add references to FlySpray items via: FlySpray#itemid - itemid is the integer id of FlySpray tracked item I tried to collect all these plus other options for DW syntax in a wiki page: http://sip-router.org/wiki/tutorials/contribute-to-dokuwiki Feel free to add more details there. Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/

1 0

[Serusers] IPTel web pages still relevant?
by SIP 03 Aug '09

03 Aug '09

A brief look at the IPTel pages show pretty much nothing but comment spam for pages and pages. Does anyone even bother to maintain the pages anymore? I'm looking for more relevant documentation to SER 2.0 (or even the SIP Router project). The docs in the doc subtree of the source for both of these projects is outdated and no longer valid with respect to any information stored in the database. It still references subscriber tables and the like. All 0.9.X format DB data. Is there an update somewhere? N.

3 3

[Kamailio-Users] Kamailio OSP-module compile problem
by Martin Werne 31 Jul '09

31 Jul '09

Hi! I have problem compiling the OSP-module in Kamailio 1.5.x OS: Ubuntu Server 8.04LTS Compiling and installing OSPToolkit is OK. The problem occures when it comes to compiling the OSP-module in Kamailio: make[1]: Entering directory `/usr/src/kamailio/modules/osp' make[1]: Leaving directory `/usr/src/kamailio/modules/osp' make[1]: Entering directory `/usr/src/kamailio/modules/osp' Compiling destination.c Compiling globals.c Compiling orig_transaction.c In file included from orig_transaction.c:40: sipheader.h:48: error: expected declaration specifiers or '...' before 'OSPTCALLID' orig_transaction.c: In function 'ospLoadRoutes': orig_transaction.c:84: error: 'OSPE_DEST_PROT' undeclared (first use in this function) orig_transaction.c:84: error: (Each undeclared identifier is reported only once orig_transaction.c:84: error: for each function it appears in.) orig_transaction.c:84: error: expected ';' before 'protocol' orig_transaction.c:85: error: 'OSPE_DEST_OSP_ENABLED' undeclared (first use in this function) orig_transaction.c:85: error: expected ';' before 'enabled' orig_transaction.c:150: error: 'protocol' undeclared (first use in this function) orig_transaction.c:154: error: 'OSPE_DEST_PROT_SIP' undeclared (first use in this function) orig_transaction.c:157: error: 'OSPE_DEST_PROT_H323_LRQ' undeclared (first use in this function) orig_transaction.c:158: error: 'OSPE_DEST_PROT_H323_SETUP' undeclared (first use in this function) orig_transaction.c:159: error: 'OSPE_DEST_PROT_IAX' undeclared (first use in this function) orig_transaction.c:163: error: 'OSPE_DEST_PROT_UNDEFINED' undeclared (first use in this function) orig_transaction.c:164: error: 'OSPE_DEST_PROT_UNKNOWN' undeclared (first use in this function) orig_transaction.c:170: error: 'enabled' undeclared (first use in this function) orig_transaction.c:174: error: 'OSPE_OSP_FALSE' undeclared (first use in this function) orig_transaction.c:189: error: 'OSPC_SOURCE' undeclared (first use in this function) orig_transaction.c: In function 'ospRequestRouting': orig_transaction.c:260: error: 'OSPTCALLID' undeclared (first use in this function) orig_transaction.c:260: error: 'callids' undeclared (first use in this function) orig_transaction.c:282: error: too many arguments to function 'ospGetCallId' orig_transaction.c:325: error: 'OSPC_E164' undeclared (first use in this function) make[1]: *** [orig_transaction.o] Error 1 make[1]: Leaving directory `/usr/src/kamailio/modules/osp' make: *** [modules] Error 2 Has tried with both OSPToolkit 3.5.2 and 3.3.6. And with both Kamailio 1.5.1 and 1.5.2 Same problem. Does anyone know what is the problem? Regards, Martin

3 4

[Kamailio-Users] About dlg_isflagset in onreply_route
by Min Wang 31 Jul '09

31 Jul '09

Hi all In kamailio 1.5.2, I used dlg_isflagset inside a ONREPLY_ROUTE. ////////////////scripts////////////////// Route[1]{ ... Dlg_manage(); Dlg_setflag("1"); ... } onreply_route[1] { #dlg_get("$ci", "$ft", "$tt"); #xlog("L_INFO", " Dialog: $dlg(h_id) . $dlg(h_entry)\n"); if (dlg_isflagset("1")) { xlog("L_INFO", " here is it \n"); } } ///////////////////////////////////////// The dlg_isflagset seems do not match anything until there is dlg_get("$ci", "$ft", "$tt"); before it. Questions are: (1) Is it the correct usage or a pontential bug. (2) How dialog is cheched/matched in the onreply_route internally? Kind Regards Min Wang

3 2

[Kamailio-Users] Parallel forking problem
by Dubravko Caric 30 Jul '09

30 Jul '09

Hi all, could you please give me suggestions how to simultaneously send call to PSTN destination for UAC that is logged in. So, when call comes for sip:bob@example.com it must also go to its PSTN destination 1234(a)10.0.0.2 (avp in mysql) and ring both destinations simultaneously. Example below doesn't work and I keep ending up with two INVITEs to PSTN GW and proxy doesn't fork call to UAC. in this case, if I understand correctly, this is proper behavior because append_branch() has its function only if UAC isn't logged in or on 480,486... And how to handle this when user is logged in? if (!lookup("location")) { ... } if(...) { avp_pushto("$ru", "$avp(s:PSTN_fwd)"); append_branch(); } Thanks a lot /dubravko

2 2

[Kamailio-Users] [ANN] Ruby XCAPClient 1.0 released
by Iñaki Baz Castillo 30 Jul '09

30 Jul '09

For those interested in building SIP and SIMPLE rich presence applications in Ruby language: http://xcapclient.rubyforge.org/ http://dev.sipdoc.net/projects/ruby-xcapclient/wiki/ Ruby XCAPClient -------------------------------------------------- A XCAP client library written in Ruby language. XCAP (RFC 4825) is a protocol on top of HTTP which allows a client (usually a VoIP/SIP device) to manipulate the contents of Presence Information Data Format (PIDF) based presence documents. These documents are stored in a server in XML format. Ruby xcapclient library implements the XCAP protocol in client side, allowing the applications to get, store, modify and delete XML documents in the server. == Features * Fetch, create/replace and delete a document. * Fetch, create/replace and delete a document element (XML node). * Fetch, create/replace and delete an element attribute. * Full configurable parameters allowing customized fields for each XCAP application, such auid, XML namespace, MIME-Type, scope (:user or :global) and default document name ("index" if not set). * Fetch the namespaces and prefixes of a document node as they are used in the server. * Manage of multiple documents per XCAP application. * Fetch the XCAP server auids, extensions and namespaces ("xcap-caps" application). * SSL. * Digest and Basic HTTP authentication. * Raise custom Ruby exception for each HTTP error response. -------------------------------------------------- Enjoy. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 5

[Kamailio-Users] rfc: xavp - extended avp
by Daniel-Constantin Mierla 29 Jul '09

29 Jul '09

Hello, some time ago I created a new branch in GIT named 'daniel/xavp' that includes code for a new pseudo-variable class 'xavp' - name being a shortcut to 'extended attribute value pair'. I collected some design details in wiki: http://sip-router.org/wiki/devel/xavp It is not a replacement for current AVP api implementation, although in the future can replace it. In the past we had couple of discussions about the limitations of avps which I tried to overcome. Here are some highlights of xavp design: - string names only - various tests showed that having the complex addressing scheme with integer id and string name does not pay off performance gains. XAVP computes and integer id for each name that is used to speed up comparison - possibility to store more data types - AVP allow string and integer data types. More can be handled by XAVP, including a generic data type where you can build your own structure and store it in the list. This is good for example to store custom structures for transaction lifetine -- right now dialog module needs to store reference to dialog structure. Optimizations can be done for any case of serial forking -- uri, dst uri, q, etc can be stored in a structure without a need to parse and build it from an encoded value stored in one or many avps - possibility to group XAVPs inside another XAVP - practically is building lists of XAVPs. This should overcome issues with naming conflicts between modules - e.g., dispatacher module can group its avps in a list named "dispatacher", lcr module in a list named 'lcrl, then there can be XAVP with name 'uri' in both lists, without the risk of misusage. Another benefit would be reducing the number of parameters, right now each module exports a parameter to set the name for each AVP, with xavp there is need for one parameter to set the name of the list, then inside that list the module is free to use predefined names without the risk of overlapping. - xavp having value integer or string can be used directly in config, whether they are in a sublist or not Check the wiki (link above) for more. Looking forward for comments, suggestions and improvements. Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/

3 4

[Kamailio-Users] debate: module interface
by Daniel-Constantin Mierla 29 Jul '09

29 Jul '09

Hello, the new sip router core and librarization of kamailio management (MI) and statistics interfaces affected the module interface. At this time, although all the fields are still in the module interface, some are passive: - mi commands exports - statistics exports - extra processes The modules in the public GIT repository use the either mod_init or mod_register to export them, so everything should work as expected. fyi: if you didn't know so far - mod_register is a function automatically called when a module is loaded, giving the possibility to export components from module in very early stage, e.g., transfomations must be exported via mod_register since they are lookep up while parsing the routing blocks -- mod_register is available in kamailio since 1.5, back ported from sip router core, you just need to implement it in the module, it is not part of module interface. Back to module interface, the three fields mislead private module implementations as everything compiles ok, but at startup the fields are not actually exported. I propose to remove them and document the transition process to new interface, which is practically couple of code lines. It this way people migrating private stuff will notice the change and update. Since we are here and we have to go through each K module and remove the fields from the module interface, maybe we can create a single interface for k and s modules. Doing a short analysis and considering the above three fields removed from k interface, then: - K interface has extra - dlopen flags - needed to specify how the .so is open, required so far by perl module, iirc - PV exports - the list of pseudo-variables exported by module - the cmd (function) exports structure has a field to specify a free fixup function (not implemented for all exported commands). The goal is to allow calling functions exported by modules for higher application languages, like perl so far, where it is needed to free the memory allocated by fixup function - S interface has extra - rpc exports - exported rpc commands - onbreak function - is this used?!? couldn't find docs about or a use case, so I consider it obsolete Then common interface would be: - name - dlopen flags - mod parameters - mod commands with free fixup - rpc commands - PV exports - mod init - child init - mod destroy - onreply callback Probably we can normalize it a bit more, comments and suggestions are welcome! Thanks, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/

1 0

[Serusers] 400 bad request of SUBSCRIBE
by 静于 29 Jul '09

29 Jul '09

hi I use pa module to support SUBSCRIBE request which Event header is "presence", and the function I used in ser.cfg is as follow : if (method=="SUBSCRIBE") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_subscription("registrar"); break; }; But the response of this request is 400 error while parsing header Server: SIP Express Route (0.9.6(i386/linux)) I find there is no place in source code to parse Server header I am I miss something in ser.cfg or any other modules should be loaded ? Could any one tell me why? ___________________________________________________________ 好玩贺卡等你发，邮箱贺卡全新上线！ http://card.mail.cn.yahoo.com/

1 0

[Kamailio-Users] Fwd: Re: RES: Can someone help me?
by Raúl Alexis Betancor Santana 28 Jul '09

28 Jul '09

Sorry Rafael, the .cfg is not big ... moreover it's a simple .cfg compared with other ones I worked on. Also I will not review/debug your config for you. Just try to do a flow analisys and check where you are wrong. With a fast read .. I could tell you that you are having troubles with the failure routes at least .. -- Raúl Alexis Betancor Santana Dimensión Virtual

1 0

Re: [Kamailio-Users] RES: Can someone help me?
by Rafael Moreno Gomes 28 Jul '09

28 Jul '09

Hi Raul Here it is, # ----------- global configuration parameters ------------------------ debug=3 fork=yes log_stderror=yes children=1 disable_tcp=no log_facility=LOG_LOCAL0 # ------------------ module loading ---------------------------------- #set module path mpath="/usr/lib/kamailio/modules/" loadmodule "sl.so" loadmodule "tm.so" modparam("tm", "fr_timer", 5) modparam("tm", "fr_inv_timer_avp", "$avp(i:704)") loadmodule "rr.so" loadmodule "usrloc.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "avpops.so" loadmodule "registrar.so" loadmodule "uri_db.so" loadmodule "db_mysql.so" loadmodule "siputils.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "acc.so" loadmodule "pv.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://root:mypass@localhost/openser") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "flags_avp", "$avp(i:712)") # ------------------------- request routing logic ------------------- route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } route(4); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); setflag(3); } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { exit; } } sl_send_reply("404","Not here"); } exit; } if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); route(3); if (!is_method("REGISTER|MESSAGE")) record_route(); if (is_method("INVITE")) { setflag(1); } if (!uri==myself) if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } setflag(2); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { if (!load_gws()) { sl_send_reply("503", "Unable to load gateways"); exit; } else { $var(i) = 0; while(is_avp_set("$(avp(i:709)[$var(i)])")) { xlog("L_INFO", "loading gw_uri_avp[$var(i)]='$(avp(i:709)[$var(i)])'\n"); $var(i) = $var(i) + 1; }; if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "trying gateway '$avp(i:709)'\n"); } else { xlog("L_INFO", "no available gateways ...\n"); }; if (next_gw()) { xlog("L_INFO", "ruri_user_avp='$avp(i:500)'\n"); t_on_failure("2"); route(1); } else { sl_send_reply("503", "No available gateways"); exit; }; }; exit; } failure_route[2] { xlog("L_INFO", "entering failure_route[2] for reply code '$T_reply_code'\n"); if (t_check_status("408|50[34]")) { if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "trying next gateway '$avp(i:709)'\n"); } else { xlog("L_INFO", "no more gateways to try ...\n"); }; if (next_gw()) { t_on_failure("2"); route(1); } else { t_reply("503", "No gateways"); exit; }; exit; }; } Rafael Moreno Matrix Telecom www.matrixtelecom.net.br / Phone: 55.11.3323-9207

1 0

[Kamailio-Users] append_branch: max nr of branches exceeded
by Ma Zhiyong 28 Jul '09

28 Jul '09

Hi, all I use kamailio 1.5.1 notls as a call-relay, It works fine but I found some errors in my logs, can you tell me what happened? Should I you increase MAX_BRANCHES value in sip-server/config.h? ERROR:core:append_branch: max nr of branches exceeded ERROR:dispatcher:ds_update_dst: append_branch action failed ERROR:dispatcher:ds_next_dst: cannot set dst addr my cfg: ds_select_domain("2","4"); force_rtp_proxy(); t_relay();

3 2

[Kamailio-Users] How to get the destination IP
by catalina oancea 28 Jul '09

28 Jul '09

Hello Is there a way to find the IP address that the request will be sent to using kamailio.cfg ? For example if extension is behind NAT, after lookup() I have in $rd the IP in the contact column of location table, not the actual IP in the received column. I couldn't find a pseudovar for this. Thanks, Catalina

3 5

[Serusers] FW: HA1 calculation failed
by jay 28 Jul '09

28 Jul '09

Hello I try to launch ser_mysql.sh as root user, When I try it using "/usr/sbin/ser_mysql.sh create" , I got the error "HA1 calculation failed" . How to overcome the problem ? Cheers, Jay

2 1

Re: [Kamailio-Users] RES: Can someone help me?
by Rafael Moreno Gomes 28 Jul '09

28 Jul '09

Hi, I have been reading a lot about Kamailio and I have done one configuration with lcr servers but now the core is giving me a message When I try: Jul 27 19:33:54 [13430] ERROR:tm:t_forward_nonack: sending request failed Jul 27 19:34:12 [13430] WARNING:core:run_actions: null action list (rec_level=2) Jul 27 19:34:12 [13430] WARNING:core:run_actions: null action list (rec_level=2) Jul 27 19:34:12 [13430] WARNING:core:run_actions: null action list (rec_level=2) Jul 27 19:34:54 [13430] WARNING:core:run_actions: null action list (rec_level=2) Jul 27 19:34:54 [13430] WARNING:core:run_actions: null action list (rec_level=2) loading gw_uri_avp[0]='1|0||3970344258||5061|2|0' loading gw_uri_avp[1]='1|0||3987121474||5061|2|0' trying gateway '1|0||3970344258||5061|2|0' ruri_user_avp='rafael' Jul 27 19:34:54 [13430] ERROR:core:tcp_blocking_connect: poll error: flags 18 Jul 27 19:34:54 [13430] ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused Jul 27 19:34:54 [13430] ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jul 27 19:34:54 [13430] ERROR:core:tcp_send: connect failed Jul 27 19:34:54 [13430] ERROR:tm:msg_send: tcp_send failed Jul 27 19:34:54 [13430] ERROR:tm:t_forward_nonack: sending request failed Rafael Moreno Matrix Telecom www.matrixtelecom.net.br / Phone: 55.11.3323-9207

2 1

[Serusers] Compile errors on CentOS 5.3
by SIP 27 Jul '09

27 Jul '09

Just grabbed the latest CVS for SER and attempted to compile it on a CentOS 5.3 box, but received these errors: config.mak loaded Makefile.defs defs skipped gcc -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev23-make"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c cfg_core.c -o cfg_core.o In file included from futexlock.h:52, from lock_ops.h:85, from locking.h:70, from pt.h:50, from cfg_core.c:42: /usr/include/linux/futex.h:108: error: expected ')' before '*' token /usr/include/linux/futex.h:112: error: expected ')' before '*' token make: *** [cfg_core.o] Error 1 As you can see, it's an X86_64 architecture box running kernel 2.6.18-128.el5. I saw a similar error in Google, but there was no resolution. Any ideas? N.

2 1

[Kamailio-Users] SIREMIS v0.9.3 Released
by Elena-Ramona Modroiu 27 Jul '09

27 Jul '09

Hi, I am happy to announce a new release of SIREMIS - the web management interface for SIP routing engines Kamailio (OpenSER) and SIP-Router.org. There are a bunch of new features, among them: - communication with SER and SIP router core via XMLRPC - communication with FreeSWITCH via event socket - rich text (html) presentation for the output of several external commands - auto-suggestion for groups management - new sortable columns and references between views For installation (upgrade) guidelines and other details, visit: http://siremis.asipto.com Some screenshots relevant for this release: http://www.asipto.com/gallery/v/siremis/siremis_20.jpg.html?g2_imageViewsIn… http://www.asipto.com/gallery/v/siremis/siremis_18.jpg.html?g2_imageViewsIn… http://www.asipto.com/gallery/v/siremis/siremis_19.jpg.html?g2_imageViewsIn… Enjoy, Ramona

1 0

[Kamailio-Users] RES: Can someone help me?
by Rafael Moreno Gomes 27 Jul '09

27 Jul '09

Ok here it goes, I have a Mera Server that I'm trying to connect my Kamailio to change my old SER server to a new level for my Retail clients, my ser will not route to any router my Mera Server will do it What I'm trying to make is a SIP:xxx.xxx.xxx.xxx:5060 on my mera server. When I register it goes ok U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-2209c854c438c347-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @192.168.5.38:63042;rinstance=bcdb08a4009ee386>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZT VkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 2 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite releas e 1103d stamp 53117..Authorization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f31e000000008e5e591d 36c39beb4f5939faad6d9d29",uri="sip:sps0.matrix.net.br" ,response="386a15e662550d45f24bccb6cbb25494",algorithm=MD5..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-2209c854c438c347-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sps0.matrix. net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d55b..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 2 R EGISTER..Contact: <sip:rafael@192.168.5.38:63042;rinstance=bcdb08a4009ee386>;expires=3600..Ser ver: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... But when I try to make calls U 200.196.28.25:63042 -> 200.196.28.124:5060 INVITE sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael@200.196.28.25:63042>..To: "551133239212"<sip:551133239212@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJ mYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 361....v=0..o=- 7 2 IN IP4 192.168.5.38..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.5.38..t=0 0..m=audio 8934 RTP/AVP 107 0 8 101..a=alt:1 3 : b2cv/jrI CAiY+XUC 192.168.5.38 8934..a=alt:2 2 : oUZIn8S0 Uq5tdU9E 192.168.52.1 8934..a=alt:3 1 : Z8/DE0ry qIvXPCKF 192.168.228.1 8934..a=fmtp:101 0-15.. a=rtpmap:107 BV32/16000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport= 63042;received=200.196.28.25..To: "55 1133239212"<sip:551133239212@sps0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb49 9b30f44b2.e355..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNT JmYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 1 INVITE..Proxy-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f32800000002a304ed8610abc47da56bdc29957a394d"..Server: Kama ilio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 ACK sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport. .To: "551133239212"<sip:5511332 39212 [!at] sps0.matrix.net.br (replace the [!at] with a @)>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.e355..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMzQ3N Dc5ZDU...CSeq: 1 ACK..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 INVITE sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael@200.196.28.25:63042>..To: "551133239212"<sip:551133239212@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJ mYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Author ization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f32800000002a304ed86 10abc47da56bdc29957a394d",uri="sip:551133239212@sps0.matrix.net.br",response ="bd0 2426b09957f78a347036f020e4c48",algorithm=MD5..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 361....v=0..o=- 7 2 IN IP4 192.168.5.38..s=CounterPath X-Lite 3.0 ..c=IN IP4 192.168.5.38..t=0 0..m=audio 8934 RTP/AVP 107 0 8 101..a=alt:1 3 : b2cv/jrI CAiY+XUC 192.168.5.38 8934..a=alt:2 2 : oUZIn8S0 Uq5tdU9E 192.168.52.1 8934..a=alt:3 1 : Z8/DE0ry qIvXPCKF 192.168.228.1 8934..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport= 63042;received=200.196.28.25..To: "551133239212"<sip:5511 33239212 [!at] sps0.matrix.net.br (replace the [!at] with a @)>;tag=bbd8f31c58bc6a3266ec5509a254d55b-2513..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMz Q3NDc5ZDU...CSeq: 2 INVITE..Server: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... Now may someone help me :D Thanks guys! ------------------------------- Message: 1 Date: Fri, 24 Jul 2009 10:56:34 +0200 From: I?aki Baz Castillo <ibc(a)aliax.net> Subject: Re: [Kamailio-Users] Can't make calls To: users(a)lists.kamailio.org Message-ID: <200907241056.34301.ibc(a)aliax.net> Content-Type: Text/Plain; charset="utf-8" El Viernes, 24 de Julio de 2009, Rafael Moreno Gomes escribi?: > Can Someone help me? No. First: Don't send the same mail twice. Second: Describe your problem in a detailed way (nobody will spend his time trying to guess which problem you have by inspecting your full configuration and SIP traces). -- I?aki Baz Castillo <ibc(a)aliax.net> ------------------------------ Message: 2 Date: Fri, 24 Jul 2009 11:43:01 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] Can't make calls To: I?aki Baz Castillo <ibc(a)aliax.net> Cc: users(a)lists.kamailio.org Message-ID: <4A698225.7080309(a)gmail.com> Content-Type: text/plain; charset=UTF-8; format=flowed On 24.07.2009 10:56 Uhr, I?aki Baz Castillo wrote: > El Viernes, 24 de Julio de 2009, Rafael Moreno Gomes escribi?: > >> Can Someone help me? >> > > No. > > First: Don't send the same mail twice. > > Second: Describe your problem in a detailed way (nobody will spend his time > trying to guess which problem you have by inspecting your full configuration > and SIP traces). > > I tried to put together some info about reporting issues. It can be used to help new people in such situation: http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:report-an-issue Please add content there, so reporting and troubleshooting becomes easier for everybody. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/index.php/sip-router-bootcamp/ ------------------------------ _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 50, Issue 49 *************************************

6 5

[Serusers] HA1 calculation failed
by jay 25 Jul '09

25 Jul '09

Hello I try to launch ser_mysql.sh as root user, When I try it using "/usr/sbin/ser_mysql.sh create" , I got the error "HA1 calculation failed" . How to overcome the problem ? Cheers, Jay

2 1

Re: [Kamailio-Users] RES: Can someone help me?
by Rafael Moreno Gomes 24 Jul '09

24 Jul '09

Ok, Now let me show what I did, I have installed the kamailio_1.5.1_i386.deb, kamailio-presence-modules_1.5.1_i386.deb, kamailio-mysql-module_1.5.1_i386.deb "Do I need to install anything more than this?" I have made the configs to kamailio work with MySQL Rafael Moreno Matrix Telecom www.matrixtelecom.net.br / Phone: 55.11.3323-9207

3 2

Re: [Kamailio-Users] RES: Can someone help me?
by Rafael Moreno Gomes 24 Jul '09

24 Jul '09

Ok, What I'm trying to do is. My Retail user connect to my SIP server by a PBX or a Softphone The SIP will make a Authentication of my user on my MySQL database. When the user make a call it need to be sent to my Whole sale server which is a MERA Server The Mera will make all the routing to my Cisco Routers and then to my PSTN or other Carriers :) Rafael Moreno Matrix Telecom www.matrixtelecom.net.br / Phone: 55.11.3323-9207

2 1

[Kamailio-Users] Can't make calls
by Rafael Moreno Gomes 24 Jul '09

24 Jul '09

Can Someone help me? ------------------------------------------------------------- NGREP LOG sps0:/# ngrep rafael(a)sps0.matrix.net.br -q interface: eth0 (200.196.28.0/255.255.255.0) match: rafael(a)sps0.matrix.net.br U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-73147a384a27ac3c-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @192.168.5.38:63042;rinstance=bcdb08a4009ee386>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZT VkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 1 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite releas e 1103d stamp 53117..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-73147a384a27ac3c-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sp s0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.07e7..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY.. .CSeq: 1 REGISTER..WWW-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f31e000000008e5e591d36c39beb4f5939faad6d9d29"..Server: Kamailio (1.5.1-notls (i386/linux)) ..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-2209c854c438c347-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @192.168.5.38:63042;rinstance=bcdb08a4009ee386>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZT VkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 2 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite releas e 1103d stamp 53117..Authorization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f31e000000008e5e591d 36c39beb4f5939faad6d9d29",uri="sip:sps0.matrix.net.br" ,response="386a15e662550d45f24bccb6cbb25494",algorithm=MD5..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-2209c854c438c347-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sps0.matrix. net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d55b..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 2 R EGISTER..Contact: <sip:rafael@192.168.5.38:63042;rinstance=bcdb08a4009ee386>;expires=3600..Ser ver: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-e552bf20e46b0448-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @192.168.5.38:63042;rinstance=bcdb08a4009ee386>;expires=0..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVk MTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 3 REGISTER..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 110 3d stamp 53117..Authorization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f31e000000008e5e591d 36c39beb4f5939faad6d9d29",uri="sip:sps0.matrix.net.br",resp onse="386a15e662550d45f24bccb6cbb25494",algorithm=MD5..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-e552bf20e46b0448-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sps0.matrix. net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.8472..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 3 R EGISTER..Server: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a95082141970153a-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @200.196.28.25:63042;rinstance=381876ea15534dcf>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1Z TVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 4 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite relea se 1103d stamp 53117..Authorization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f31e000000008e5e591d 36c39beb4f5939faad6d9d29",uri="sip:sps0.matrix.net.br ",response="386a15e662550d45f24bccb6cbb25494",algorithm=MD5..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a95082141970153a-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sps0.matrix. net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.26f5..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 4 R EGISTER..Contact: <sip:rafael@200.196.28.25:63042;rinstance=381876ea15534dcf>;expires=3600..Se rver: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 SUBSCRIBE sip:rafael@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-7c5a2623fb530542-1---d8754z-;rport. .Max-Forwards: 70..Contact: <si p:rafael@200.196.28.25:63042>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVk YWIyZmMwM2Q...CSeq: 1 SUBSCRIBE..Expires: 300..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1103d stamp 5311 7..Event: message-summary..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-7c5a2623fb530542-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SP S0"<sip:rafael@sps0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.39a7 ..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJm YzVkYWIyZmMwM2Q...CSeq: 1 SUBSCRIBE..Proxy-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f321000000012f11dfcfba5649b1bb6b23b82b0749a9"..Server: Kamailio (1.5.1 -notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 SUBSCRIBE sip:rafael@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-59288923603d7744-1---d8754z-;rport. .Max-Forwards: 70..Contact: <si p:rafael@200.196.28.25:63042>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVk YWIyZmMwM2Q...CSeq: 2 SUBSCRIBE..Expires: 300..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Proxy-Authorization: Digest username="rafae l",realm="sps0.matrix.net.br",nonce="4a68f321000000012f11dfcfba5649b1bb6b23b 82b0749a9",uri="sip:rafael@sps0.matrix.net.br",response="ea373d573d66ba13867 11d34d96d75c1",algori thm=MD5..User-Agent: X-Lite release 1103d stamp 53117..Event: message-summary..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SUBSCRIBE sip:rafael@200.196.28.25:63042;rinstance=381876ea15534dcf SIP/2.0..Record-Route: <sip:200.196.28.124;lr=on>..Via: SIP/2.0/UDP 200.196.28.124;branch=z9hG4bK3224.45a 47c61.0..Via: SIP/2.0/UDP 192.168.5.38:63042;received=200.196.28.25;branch=z9hG4bK-d8754z-59288923603d 7744-1---d8754z-;rport=63042..Max-Forwards: 69..Contact: <sip:rafael@20 0.196.28.25:63042>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVkYWIyZmMwM2Q ...CSeq: 2 SUBSCRIBE..Expires: 300..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1103d stamp 53117..Event: m essage-summary..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 SIP/2.0 489 Event Package Not Supported..Via: SIP/2.0/UDP 200.196.28.124;branch=z9hG4bK3224.45a47c61.0..Via: SIP/2.0/UDP 192.168.5.38:63042;received=200.196.28.25;branch=z9h G4bK-d8754z-59288923603d7744-1---d8754z-;rport=63042..To: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=be58b818..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call- ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVkYWIyZmMwM2Q...CSeq: 2 SUBSCRIBE..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 489 Event Package Not Supported..Via: SIP/2.0/UDP 192.168.5.38:63042;received=200.196.28.25;branch=z9hG4bK-d8754z-59288923603d 7744-1---d8754z-;rport=63042..To: "SPS0 "<sip:rafael@sps0.matrix.net.br>;tag=be58b818..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVkYWIyZmMwM2Q...CSeq: 2 SUBSC RIBE..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 INVITE sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael@200.196.28.25:63042>..To: "551133239212"<sip:551133239212@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJ mYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 361....v=0..o=- 7 2 IN IP4 192.168.5.38..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.5.38..t=0 0..m=audio 8934 RTP/AVP 107 0 8 101..a=alt:1 3 : b2cv/jrI CAiY+XUC 192.168.5.38 8934..a=alt:2 2 : oUZIn8S0 Uq5tdU9E 192.168.52.1 8934..a=alt:3 1 : Z8/DE0ry qIvXPCKF 192.168.228.1 8934..a=fmtp:101 0-15.. a=rtpmap:107 BV32/16000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport= 63042;received=200.196.28.25..To: "55 1133239212"<sip:551133239212@sps0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb49 9b30f44b2.e355..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNT JmYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 1 INVITE..Proxy-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f32800000002a304ed8610abc47da56bdc29957a394d"..Server: Kama ilio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 ACK sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport. .To: "551133239212"<sip:5511332 39212(a)sps0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.e355..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMzQ3N Dc5ZDU...CSeq: 1 ACK..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 INVITE sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael@200.196.28.25:63042>..To: "551133239212"<sip:551133239212@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJ mYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Author ization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f32800000002a304ed86 10abc47da56bdc29957a394d",uri="sip:551133239212@sps0.matrix.net.br",response ="bd0 2426b09957f78a347036f020e4c48",algorithm=MD5..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 361....v=0..o=- 7 2 IN IP4 192.168.5.38..s=CounterPath X-Lite 3.0 ..c=IN IP4 192.168.5.38..t=0 0..m=audio 8934 RTP/AVP 107 0 8 101..a=alt:1 3 : b2cv/jrI CAiY+XUC 192.168.5.38 8934..a=alt:2 2 : oUZIn8S0 Uq5tdU9E 192.168.52.1 8934..a=alt:3 1 : Z8/DE0ry qIvXPCKF 192.168.228.1 8934..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport= 63042;received=200.196.28.25..To: "551133239212"<sip:5511 33239212(a)sps0.matrix.net.br>;tag=bbd8f31c58bc6a3266ec5509a254d55b-2513..From : "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMz Q3NDc5ZDU...CSeq: 2 INVITE..Server: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 ACK sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport. .To: "551133239212"<sip:5511332 39212(a)sps0.matrix.net.br>;tag=bbd8f31c58bc6a3266ec5509a254d55b-2513..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMzQ3N Dc5ZDU...CSeq: 2 ACK..Content-Length: 0.... ------------------------------------------------------------- SIP LOG SIP/2.0 489 Event Package Not Supported Via: SIP/2.0/UDP 192.168.5.38:63042;received=200.196.28.25;branch=z9hG4bK-d8754z-59288923603d 7744-1---d8754z-;rport=63042 To: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=be58b818 From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236 Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVkYWIyZmMwM2Q. CSeq: 2 SUBSCRIBE User-Agent: X-Lite release 1103d stamp 53117 Content-Length: 0 Jul 23 20:32:19 [2948] DBG:core:_shm_resize: resize(0) called Jul 23 20:32:19 [2948] DBG:tm:insert_timer_unsafe: [2]: 0xafc1d4f8 (15) Jul 23 20:32:19 [2948] DBG:tm:relay_reply: sent buf=0x8149d48: SIP/2.0 4..., shmem=0xafc1f430: SIP/2.0 4 Jul 23 20:32:19 [2948] DBG:tm:run_trans_callbacks: trans=0xafc1d4b0, callback type 256, id 0 entered Jul 23 20:32:19 [2948] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Jul 23 20:32:19 [2948] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jul 23 20:32:19 [2948] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:19 [2948] DBG:core:receive_msg: cleaning up Jul 23 20:32:20 [2950] DBG:tm:utimer_routine: timer routine:4,tl=0xafc1d5fc next=(nil), timeout=10400000 Jul 23 20:32:24 [2950] DBG:tm:timer_routine: timer routine:2,tl=0xafc1d4f8 next=(nil), timeout=15 Jul 23 20:32:24 [2950] DBG:tm:wait_handler: removing 0xafc1d4b0 from table Jul 23 20:32:24 [2950] DBG:tm:delete_cell: delete transaction 0xafc1d4b0 Jul 23 20:32:24 [2950] DBG:tm:wait_handler: done Jul 23 20:32:25 [2946] DBG:core:udp_rcv_loop: probing packet received from 200.202.22.5 55325 Jul 23 20:32:26 [2949] DBG:core:parse_msg: SIP Request: Jul 23 20:32:26 [2949] DBG:core:parse_msg: method: <INVITE> Jul 23 20:32:26 [2949] DBG:core:parse_msg: uri: <sip:551133239212@sps0.matrix.net.br> Jul 23 20:32:26 [2949] DBG:core:parse_msg: version: <SIP/2.0> Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=2 Jul 23 20:32:26 [2949] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-a957a2771c264219-1---d8754z->; state=6 Jul 23 20:32:26 [2949] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jul 23 20:32:26 [2949] DBG:core:parse_via: end of header reached, state=5 Jul 23 20:32:26 [2949] DBG:core:parse_headers: via found, flags=2 Jul 23 20:32:26 [2949] DBG:core:parse_headers: this is the first via Jul 23 20:32:26 [2949] DBG:core:receive_msg: After parse_msg... Jul 23 20:32:26 [2949] DBG:core:receive_msg: preparing to run routing scripts... Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=100 Jul 23 20:32:26 [2949] DBG:maxfwd:is_maxfwd_present: value = 70 Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=8 Jul 23 20:32:26 [2949] DBG:core:parse_to: end of header reached, state=10 Jul 23 20:32:26 [2949] DBG:core:parse_to: display={"551133239212"}, ruri={sip:551133239212@sps0.matrix.net.br} Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: <To> [53]; uri=[sip:551133239212@sps0.matrix.net.br] Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: to body ["551133239212"<sip:551133239212@sps0.matrix.net.br> ] Jul 23 20:32:26 [2949] DBG:siputils:has_totag: no totag Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=78 Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: cseq <CSeq>: <1> <INVITE> Jul 23 20:32:26 [2949] DBG:tm:t_lookup_request: start searching: hash=24133, isACK=0 Jul 23 20:32:26 [2949] DBG:tm:matching_3261: RFC3261 transaction matching failed Jul 23 20:32:26 [2949] DBG:tm:t_lookup_request: no transaction found Jul 23 20:32:26 [2949] DBG:core:parse_to_param: tag=b45cd830 Jul 23 20:32:26 [2949] DBG:core:parse_to: end of header reached, state=29 Jul 23 20:32:26 [2949] DBG:core:parse_to: display={"SPS0"}, ruri={sip:rafael@sps0.matrix.net.br} Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: no match for: [sps0.matrix.net.br:5060] Jul 23 20:32:26 [2949] DBG:core:grep_aliases: comparing host [0:sps0.matrix.net.br:5060] with us [2:sps0.matrix.net.br:5060] Jul 23 20:32:26 [2949] DBG:core:grep_aliases: match found for: [0:sps0.matrix.net.br:5060] Jul 23 20:32:26 [2949] DBG:core:check_self: host == me Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=10000 Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: content_length=361 Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: found end of header Jul 23 20:32:26 [2949] DBG:auth:pre_auth: credentials with given realm not found Jul 23 20:32:26 [2949] DBG:auth:reserve_nonce_index: second= 16, sec_monit= -1, index= 2 Jul 23 20:32:26 [2949] DBG:auth:build_auth_hf: nonce index= 2 Jul 23 20:32:26 [2949] DBG:auth:build_auth_hf: 'Proxy-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f32800000002a304ed8610abc47da56bdc29957a394d" ' Jul 23 20:32:26 [2949] DBG:sl:send_reply: reply in stateless mode (sl) Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=ffffffffffffffff Jul 23 20:32:26 [2949] DBG:core:check_via_address: params 200.196.28.25, 192.168.5.38, 0 Jul 23 20:32:26 [2949] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:26 [2949] DBG:core:receive_msg: cleaning up Jul 23 20:32:26 [2947] DBG:core:parse_msg: SIP Request: Jul 23 20:32:26 [2947] DBG:core:parse_msg: method: <ACK> Jul 23 20:32:26 [2947] DBG:core:parse_msg: uri: <sip:551133239212@sps0.matrix.net.br> Jul 23 20:32:26 [2947] DBG:core:parse_msg: version: <SIP/2.0> Jul 23 20:32:26 [2947] DBG:core:parse_headers: flags=2 Jul 23 20:32:26 [2947] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-a957a2771c264219-1---d8754z->; state=6 Jul 23 20:32:26 [2947] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jul 23 20:32:26 [2947] DBG:core:parse_via: end of header reached, state=5 Jul 23 20:32:26 [2947] DBG:core:parse_headers: via found, flags=2 Jul 23 20:32:26 [2947] DBG:core:parse_headers: this is the first via Jul 23 20:32:26 [2947] DBG:core:receive_msg: After parse_msg... Jul 23 20:32:26 [2947] DBG:core:receive_msg: preparing to run routing scripts... Jul 23 20:32:26 [2947] DBG:core:parse_headers: flags=8 Jul 23 20:32:26 [2947] DBG:core:parse_to_param: tag=f8f2ab2c1295e90ed7dbb499b30f44b2.e355 Jul 23 20:32:26 [2947] DBG:core:parse_to: end of header reached, state=29 Jul 23 20:32:26 [2947] DBG:core:parse_to: display={"551133239212"}, ruri={sip:551133239212@sps0.matrix.net.br} Jul 23 20:32:26 [2947] DBG:core:get_hdr_field: <To> [95]; uri=[sip:551133239212@sps0.matrix.net.br] Jul 23 20:32:26 [2947] DBG:core:get_hdr_field: to body ["551133239212"<sip:551133239212@sps0.matrix.net.br>] Jul 23 20:32:26 [2947] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Jul 23 20:32:26 [2947] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:26 [2947] DBG:core:receive_msg: cleaning up Jul 23 20:32:27 [2946] DBG:core:parse_msg: SIP Request: Jul 23 20:32:27 [2946] DBG:core:parse_msg: method: <INVITE> Jul 23 20:32:27 [2946] DBG:core:parse_msg: uri: <sip:551133239212@sps0.matrix.net.br> Jul 23 20:32:27 [2946] DBG:core:parse_msg: version: <SIP/2.0> Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=2 Jul 23 20:32:27 [2946] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-267c082e8463a77a-1---d8754z->; state=6 Jul 23 20:32:27 [2946] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jul 23 20:32:27 [2946] DBG:core:parse_via: end of header reached, state=5 Jul 23 20:32:27 [2946] DBG:core:parse_headers: via found, flags=2 Jul 23 20:32:27 [2946] DBG:core:parse_headers: this is the first via Jul 23 20:32:27 [2946] DBG:core:receive_msg: After parse_msg... Jul 23 20:32:27 [2946] DBG:core:receive_msg: preparing to run routing scripts... Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=100 Jul 23 20:32:27 [2946] DBG:maxfwd:is_maxfwd_present: value = 70 Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=8 Jul 23 20:32:27 [2946] DBG:core:parse_to: end of header reached, state=10 Jul 23 20:32:27 [2946] DBG:core:parse_to: display={"551133239212"}, ruri={sip:551133239212@sps0.matrix.net.br} Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: <To> [53]; uri=[sip:551133239212@sps0.matrix.net.br] Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: to body ["551133239212"<sip:551133239212@sps0.matrix.net.br> ] Jul 23 20:32:27 [2946] DBG:siputils:has_totag: no totag Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=78 Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Jul 23 20:32:27 [2946] DBG:tm:t_lookup_request: start searching: hash=24130, isACK=0 Jul 23 20:32:27 [2946] DBG:tm:matching_3261: RFC3261 transaction matching failed Jul 23 20:32:27 [2946] DBG:tm:t_lookup_request: no transaction found Jul 23 20:32:27 [2946] DBG:core:parse_to_param: tag=b45cd830 Jul 23 20:32:27 [2946] DBG:core:parse_to: end of header reached, state=29 Jul 23 20:32:27 [2946] DBG:core:parse_to: display={"SPS0"}, ruri={sip:rafael@sps0.matrix.net.br} Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: no match for: [sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:grep_aliases: comparing host [0:sps0.matrix.net.br:5060] with us [2:sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:grep_aliases: match found for: [0:sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:check_self: host == me Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=10000 Jul 23 20:32:27 [2946] DBG:auth:check_nonce: comparing [4a68f32800000002a304ed8610abc47da56bdc29957a394d] and [4a68f32800000002a304ed8610abc47da56bdc29957a394d] Jul 23 20:32:27 [2946] DBG:core:db_new_result: allocate 28 bytes for result set at 0x814db68 Jul 23 20:32:27 [2946] DBG:db_mysql:db_mysql_get_columns: 1 columns returned from the query Jul 23 20:32:27 [2946] DBG:core:db_allocate_columns: allocate 4 bytes for result names at 0x8145e18 Jul 23 20:32:27 [2946] DBG:core:db_allocate_columns: allocate 4 bytes for result types at 0x8149f50 Jul 23 20:32:27 [2946] DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[0] at 0x8149798 Jul 23 20:32:27 [2946] DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x8149798)[0]=[password] Jul 23 20:32:27 [2946] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Jul 23 20:32:27 [2946] DBG:core:db_allocate_rows: allocate 8 bytes for rows at 0x8149c70 Jul 23 20:32:27 [2946] DBG:core:db_allocate_row: allocate 20 bytes for row values at 0x813e0c0 Jul 23 20:32:27 [2946] DBG:core:db_str2val: converting STRING [rmg25788] Jul 23 20:32:27 [2946] DBG:auth_db:get_ha1: HA1 string calculated: 25eb28681da92677e1e02a0e83d6f255 Jul 23 20:32:27 [2946] DBG:auth:check_response: our result = 'bd02426b09957f78a347036f020e4c48' Jul 23 20:32:27 [2946] DBG:auth:check_response: authorization is OK Jul 23 20:32:27 [2946] DBG:auth:post_auth: nonce index= 2 Jul 23 20:32:27 [2946] DBG:core:db_free_columns: freeing 1 columns Jul 23 20:32:27 [2946] DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x8149798 Jul 23 20:32:27 [2946] DBG:core:db_free_columns: freeing result names at 0x8145e18 Jul 23 20:32:27 [2946] DBG:core:db_free_columns: freeing result types at 0x8149f50 Jul 23 20:32:27 [2946] DBG:core:db_free_rows: freeing 1 rows Jul 23 20:32:27 [2946] DBG:core:db_free_row: freeing row values at 0x813e0c0 Jul 23 20:32:27 [2946] DBG:core:db_free_rows: freeing rows at 0x8149c70 Jul 23 20:32:27 [2946] DBG:core:db_free_result: freeing result set at 0x814db68 Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=14000 Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: content_length=361 Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: found end of header Jul 23 20:32:27 [2946] DBG:core:comp_scriptvar: str 29 : rafael Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: no match for: [sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:grep_aliases: comparing host [0:sps0.matrix.net.br:5060] with us [2:sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:grep_aliases: match found for: [0:sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:check_self: host == me Jul 23 20:32:27 [2946] DBG:registrar:lookup: '551133239212' Not found in usrloc Jul 23 20:32:27 [2946] DBG:tm:t_newtran: transaction on entrance=(nil) Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=ffffffffffffffff Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=78 Jul 23 20:32:27 [2946] DBG:tm:t_lookup_request: start searching: hash=24130, isACK=0 Jul 23 20:32:27 [2946] DBG:tm:matching_3261: RFC3261 transaction matching failed Jul 23 20:32:27 [2946] DBG:tm:t_lookup_request: no transaction found Jul 23 20:32:27 [2946] DBG:tm:run_reqin_callbacks: trans=0xafc1d4b0, callback type 1, id 0 entered Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=78 Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=ffffffffffffffff Jul 23 20:32:27 [2946] DBG:core:check_via_address: params 200.196.28.25, 192.168.5.38, 0 Jul 23 20:32:27 [2946] DBG:core:_shm_resize: resize(0) called Jul 23 20:32:27 [2946] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Jul 23 20:32:27 [2946] DBG:tm:set_timer: relative timeout is 500000 Jul 23 20:32:27 [2946] DBG:tm:insert_timer_unsafe: [4]: 0xafc1d578 (17700000) Jul 23 20:32:27 [2946] DBG:tm:set_timer: relative timeout is 30 Jul 23 20:32:27 [2946] DBG:tm:insert_timer_unsafe: [0]: 0xafc1d594 (47) Jul 23 20:32:27 [2946] DBG:tm:_reply_light: reply sent out. buf=0x814ddf8: SIP/2.0 4..., shmem=0xafc1f148: SIP/2.0 4 Jul 23 20:32:27 [2946] DBG:tm:run_trans_callbacks: trans=0xafc1d4b0, callback type 256, id 0 entered Jul 23 20:32:27 [2946] DBG:tm:_reply_light: finished Jul 23 20:32:27 [2946] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jul 23 20:32:27 [2946] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:27 [2946] DBG:core:receive_msg: cleaning up Jul 23 20:32:27 [2948] DBG:core:parse_msg: SIP Request: Jul 23 20:32:27 [2948] DBG:core:parse_msg: method: <ACK> Jul 23 20:32:27 [2948] DBG:core:parse_msg: uri: <sip:551133239212@sps0.matrix.net.br> Jul 23 20:32:27 [2948] DBG:core:parse_msg: version: <SIP/2.0> Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=2 Jul 23 20:32:27 [2948] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-267c082e8463a77a-1---d8754z->; state=6 Jul 23 20:32:27 [2948] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jul 23 20:32:27 [2948] DBG:core:parse_via: end of header reached, state=5 Jul 23 20:32:27 [2948] DBG:core:parse_headers: via found, flags=2 Jul 23 20:32:27 [2948] DBG:core:parse_headers: this is the first via Jul 23 20:32:27 [2948] DBG:core:receive_msg: After parse_msg... Jul 23 20:32:27 [2948] DBG:core:receive_msg: preparing to run routing scripts... Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=8 Jul 23 20:32:27 [2948] DBG:core:parse_to_param: tag=bbd8f31c58bc6a3266ec5509a254d55b-2513 Jul 23 20:32:27 [2948] DBG:core:parse_to: end of header reached, state=29 Jul 23 20:32:27 [2948] DBG:core:parse_to: display={"551133239212"}, ruri={sip:551133239212@sps0.matrix.net.br} Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: <To> [95]; uri=[sip:551133239212@sps0.matrix.net.br] Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: to body ["551133239212"<sip:551133239212@sps0.matrix.net.br>] Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=100 Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: cseq <CSeq>: <2> <ACK> Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: content_length=0 Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: found end of header Jul 23 20:32:27 [2948] DBG:maxfwd:is_maxfwd_present: max_forwards header not found! Jul 23 20:32:27 [2948] DBG:siputils:has_totag: totag found Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=200 Jul 23 20:32:27 [2948] DBG:rr:find_first_route: No Route headers found Jul 23 20:32:27 [2948] DBG:rr:loose_route: There is no Route HF Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=78 Jul 23 20:32:27 [2948] DBG:tm:t_lookup_request: start searching: hash=24130, isACK=1 Jul 23 20:32:27 [2948] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d8754z-267c082e8463a77a-1---d8754z- Jul 23 20:32:27 [2948] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Jul 23 20:32:27 [2948] DBG:tm:t_lookup_request: transaction found (T=0xafc1d4b0) Jul 23 20:32:27 [2948] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Jul 23 20:32:27 [2948] DBG:tm:insert_timer_unsafe: [2]: 0xafc1d4f8 (22) Jul 23 20:32:27 [2948] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jul 23 20:32:27 [2948] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:27 [2948] DBG:core:receive_msg: cleaning up Jul 23 20:32:27 [2950] DBG:tm:utimer_routine: timer routine:4,tl=0xafc1d578 next=(nil), timeout=17700000 Jul 23 20:32:31 [2950] DBG:tm:timer_routine: timer routine:2,tl=0xafc1d4f8 next=(nil), timeout=22 Jul 23 20:32:31 [2950] DBG:tm:wait_handler: removing 0xafc1d4b0 from table Jul 23 20:32:31 [2950] DBG:tm:delete_cell: delete transaction 0xafc1d4b0 Jul 23 20:32:31 [2950] DBG:tm:wait_handler: done Jul 23 20:32:34 [2947] DBG:core:udp_rcv_loop: probing packet received from 200.202.22.5 55325 ------------------------------------------------------------- My Config File # # $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=7 log_stderror=yes log_facility=LOG_LOCAL7 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:mypassword@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://root:mypassword@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE) #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ------------------------------------------------------------- kamctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 3 | +1 | | 1 | 1 | | 4 | +55 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateways +---------+----------------+----------+------+------------+-----------+----- ---+-------+------+-------+--------+------+ | gw_name | ip_addr | hostname | port | uri_scheme | transport | grp_id | strip | tag | flags | weight | ping | +---------+----------------+----------+------+------------+-----------+----- ---+-------+------+-------+--------+------+ | MERAI | 66.165.166.236 | | 5061 | 1 | 2 | 1 | 0 | | 0 | 1 | 0 | | MERAII | 66.165.166.237 | | 5061 | 1 | 2 | 1 | 0 | | 0 | 1 | 0 | +---------+----------------+----------+------+------------+-----------+----- ---+-------+------+-------+--------+------+ ------------------------------------------------------------- kamctl dispatcher show dispatcher gateways +----+-------+-------------------------+-------+----------+-------------+ | id | setid | destination | flags | priority | description | +----+-------+-------------------------+-------+----------+-------------+ | 2 | 1 | sip:66.165.166.236:5061 | 1 | 1 | MERA | | 3 | 2 | sip:66.165.166.237:5061 | 3 | 0 | MERAII | +----+-------+-------------------------+-------+----------+-------------+ Rafael Moreno Matrix Telecom www.matrixtelecom.net.br / Phone: 55.11.3323-9207

3 2

[Kamailio-Users] Can't make calls
by Rafael Moreno Gomes 23 Jul '09

23 Jul '09

Can Someone help me? ------------------------------------------------------------- NGREP LOG sps0:/# ngrep rafael(a)sps0.matrix.net.br -q interface: eth0 (200.196.28.0/255.255.255.0) match: rafael(a)sps0.matrix.net.br U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-73147a384a27ac3c-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @192.168.5.38:63042;rinstance=bcdb08a4009ee386>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZT VkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 1 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite releas e 1103d stamp 53117..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-73147a384a27ac3c-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sp s0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.07e7..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY.. .CSeq: 1 REGISTER..WWW-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f31e000000008e5e591d36c39beb4f5939faad6d9d29"..Server: Kamailio (1.5.1-notls (i386/linux)) ..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-2209c854c438c347-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @192.168.5.38:63042;rinstance=bcdb08a4009ee386>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZT VkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 2 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite releas e 1103d stamp 53117..Authorization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f31e000000008e5e591d 36c39beb4f5939faad6d9d29",uri="sip:sps0.matrix.net.br" ,response="386a15e662550d45f24bccb6cbb25494",algorithm=MD5..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-2209c854c438c347-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sps0.matrix. net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d55b..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 2 R EGISTER..Contact: <sip:rafael@192.168.5.38:63042;rinstance=bcdb08a4009ee386>;expires=3600..Ser ver: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-e552bf20e46b0448-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @192.168.5.38:63042;rinstance=bcdb08a4009ee386>;expires=0..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVk MTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 3 REGISTER..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 110 3d stamp 53117..Authorization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f31e000000008e5e591d 36c39beb4f5939faad6d9d29",uri="sip:sps0.matrix.net.br",resp onse="386a15e662550d45f24bccb6cbb25494",algorithm=MD5..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-e552bf20e46b0448-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sps0.matrix. net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.8472..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 3 R EGISTER..Server: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a95082141970153a-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael @200.196.28.25:63042;rinstance=381876ea15534dcf>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1Z TVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 4 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite relea se 1103d stamp 53117..Authorization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f31e000000008e5e591d 36c39beb4f5939faad6d9d29",uri="sip:sps0.matrix.net.br ",response="386a15e662550d45f24bccb6cbb25494",algorithm=MD5..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a95082141970153a-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sps0.matrix. net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.26f5..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID: ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 4 R EGISTER..Contact: <sip:rafael@200.196.28.25:63042;rinstance=381876ea15534dcf>;expires=3600..Se rver: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 SUBSCRIBE sip:rafael@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-7c5a2623fb530542-1---d8754z-;rport. .Max-Forwards: 70..Contact: <si p:rafael@200.196.28.25:63042>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVk YWIyZmMwM2Q...CSeq: 1 SUBSCRIBE..Expires: 300..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1103d stamp 5311 7..Event: message-summary..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-7c5a2623fb530542-1---d8754z-;rport= 63042;received=200.196.28.25..To: "SP S0"<sip:rafael@sps0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.39a7 ..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJm YzVkYWIyZmMwM2Q...CSeq: 1 SUBSCRIBE..Proxy-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f321000000012f11dfcfba5649b1bb6b23b82b0749a9"..Server: Kamailio (1.5.1 -notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 SUBSCRIBE sip:rafael@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-59288923603d7744-1---d8754z-;rport. .Max-Forwards: 70..Contact: <si p:rafael@200.196.28.25:63042>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVk YWIyZmMwM2Q...CSeq: 2 SUBSCRIBE..Expires: 300..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Proxy-Authorization: Digest username="rafae l",realm="sps0.matrix.net.br",nonce="4a68f321000000012f11dfcfba5649b1bb6b23b 82b0749a9",uri="sip:rafael@sps0.matrix.net.br",response="ea373d573d66ba13867 11d34d96d75c1",algori thm=MD5..User-Agent: X-Lite release 1103d stamp 53117..Event: message-summary..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SUBSCRIBE sip:rafael@200.196.28.25:63042;rinstance=381876ea15534dcf SIP/2.0..Record-Route: <sip:200.196.28.124;lr=on>..Via: SIP/2.0/UDP 200.196.28.124;branch=z9hG4bK3224.45a 47c61.0..Via: SIP/2.0/UDP 192.168.5.38:63042;received=200.196.28.25;branch=z9hG4bK-d8754z-59288923603d 7744-1---d8754z-;rport=63042..Max-Forwards: 69..Contact: <sip:rafael@20 0.196.28.25:63042>..To: "SPS0"<sip:rafael@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVkYWIyZmMwM2Q ...CSeq: 2 SUBSCRIBE..Expires: 300..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1103d stamp 53117..Event: m essage-summary..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 SIP/2.0 489 Event Package Not Supported..Via: SIP/2.0/UDP 200.196.28.124;branch=z9hG4bK3224.45a47c61.0..Via: SIP/2.0/UDP 192.168.5.38:63042;received=200.196.28.25;branch=z9h G4bK-d8754z-59288923603d7744-1---d8754z-;rport=63042..To: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=be58b818..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call- ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVkYWIyZmMwM2Q...CSeq: 2 SUBSCRIBE..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 0.... U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 489 Event Package Not Supported..Via: SIP/2.0/UDP 192.168.5.38:63042;received=200.196.28.25;branch=z9hG4bK-d8754z-59288923603d 7744-1---d8754z-;rport=63042..To: "SPS0 "<sip:rafael@sps0.matrix.net.br>;tag=be58b818..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236..Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVkYWIyZmMwM2Q...CSeq: 2 SUBSC RIBE..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 INVITE sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael@200.196.28.25:63042>..To: "551133239212"<sip:551133239212@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJ mYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 361....v=0..o=- 7 2 IN IP4 192.168.5.38..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.5.38..t=0 0..m=audio 8934 RTP/AVP 107 0 8 101..a=alt:1 3 : b2cv/jrI CAiY+XUC 192.168.5.38 8934..a=alt:2 2 : oUZIn8S0 Uq5tdU9E 192.168.52.1 8934..a=alt:3 1 : Z8/DE0ry qIvXPCKF 192.168.228.1 8934..a=fmtp:101 0-15.. a=rtpmap:107 BV32/16000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport= 63042;received=200.196.28.25..To: "55 1133239212"<sip:551133239212@sps0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb49 9b30f44b2.e355..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNT JmYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 1 INVITE..Proxy-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f32800000002a304ed8610abc47da56bdc29957a394d"..Server: Kama ilio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 ACK sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport. .To: "551133239212"<sip:5511332 39212(a)sps0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.e355..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMzQ3N Dc5ZDU...CSeq: 1 ACK..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 INVITE sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport. .Max-Forwards: 70..Contact: <sip:rafael@200.196.28.25:63042>..To: "551133239212"<sip:551133239212@sps0.matrix.net.br>..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJ mYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Author ization: Digest username="rafael",realm="sps0.matrix.net.br",nonce="4a68f32800000002a304ed86 10abc47da56bdc29957a394d",uri="sip:551133239212@sps0.matrix.net.br",response ="bd0 2426b09957f78a347036f020e4c48",algorithm=MD5..User-Agent: X-Lite release 1103d stamp 53117..Content-Length: 361....v=0..o=- 7 2 IN IP4 192.168.5.38..s=CounterPath X-Lite 3.0 ..c=IN IP4 192.168.5.38..t=0 0..m=audio 8934 RTP/AVP 107 0 8 101..a=alt:1 3 : b2cv/jrI CAiY+XUC 192.168.5.38 8934..a=alt:2 2 : oUZIn8S0 Uq5tdU9E 192.168.52.1 8934..a=alt:3 1 : Z8/DE0ry qIvXPCKF 192.168.228.1 8934..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. U 200.196.28.124:5060 -> 200.196.28.25:63042 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport= 63042;received=200.196.28.25..To: "551133239212"<sip:5511 33239212(a)sps0.matrix.net.br>;tag=bbd8f31c58bc6a3266ec5509a254d55b-2513..From : "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMz Q3NDc5ZDU...CSeq: 2 INVITE..Server: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0.... U 200.196.28.25:63042 -> 200.196.28.124:5060 ACK sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP 192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport. .To: "551133239212"<sip:5511332 39212(a)sps0.matrix.net.br>;tag=bbd8f31c58bc6a3266ec5509a254d55b-2513..From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID: NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMzQ3N Dc5ZDU...CSeq: 2 ACK..Content-Length: 0.... ------------------------------------------------------------- SIP LOG SIP/2.0 489 Event Package Not Supported Via: SIP/2.0/UDP 192.168.5.38:63042;received=200.196.28.25;branch=z9hG4bK-d8754z-59288923603d 7744-1---d8754z-;rport=63042 To: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=be58b818 From: "SPS0"<sip:rafael@sps0.matrix.net.br>;tag=d1708236 Call-ID: ZjdlZWUwNjkzM2ZmYzk1MDBkMGJmYzVkYWIyZmMwM2Q. CSeq: 2 SUBSCRIBE User-Agent: X-Lite release 1103d stamp 53117 Content-Length: 0 Jul 23 20:32:19 [2948] DBG:core:_shm_resize: resize(0) called Jul 23 20:32:19 [2948] DBG:tm:insert_timer_unsafe: [2]: 0xafc1d4f8 (15) Jul 23 20:32:19 [2948] DBG:tm:relay_reply: sent buf=0x8149d48: SIP/2.0 4..., shmem=0xafc1f430: SIP/2.0 4 Jul 23 20:32:19 [2948] DBG:tm:run_trans_callbacks: trans=0xafc1d4b0, callback type 256, id 0 entered Jul 23 20:32:19 [2948] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Jul 23 20:32:19 [2948] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jul 23 20:32:19 [2948] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:19 [2948] DBG:core:receive_msg: cleaning up Jul 23 20:32:20 [2950] DBG:tm:utimer_routine: timer routine:4,tl=0xafc1d5fc next=(nil), timeout=10400000 Jul 23 20:32:24 [2950] DBG:tm:timer_routine: timer routine:2,tl=0xafc1d4f8 next=(nil), timeout=15 Jul 23 20:32:24 [2950] DBG:tm:wait_handler: removing 0xafc1d4b0 from table Jul 23 20:32:24 [2950] DBG:tm:delete_cell: delete transaction 0xafc1d4b0 Jul 23 20:32:24 [2950] DBG:tm:wait_handler: done Jul 23 20:32:25 [2946] DBG:core:udp_rcv_loop: probing packet received from 200.202.22.5 55325 Jul 23 20:32:26 [2949] DBG:core:parse_msg: SIP Request: Jul 23 20:32:26 [2949] DBG:core:parse_msg: method: <INVITE> Jul 23 20:32:26 [2949] DBG:core:parse_msg: uri: <sip:551133239212@sps0.matrix.net.br> Jul 23 20:32:26 [2949] DBG:core:parse_msg: version: <SIP/2.0> Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=2 Jul 23 20:32:26 [2949] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-a957a2771c264219-1---d8754z->; state=6 Jul 23 20:32:26 [2949] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jul 23 20:32:26 [2949] DBG:core:parse_via: end of header reached, state=5 Jul 23 20:32:26 [2949] DBG:core:parse_headers: via found, flags=2 Jul 23 20:32:26 [2949] DBG:core:parse_headers: this is the first via Jul 23 20:32:26 [2949] DBG:core:receive_msg: After parse_msg... Jul 23 20:32:26 [2949] DBG:core:receive_msg: preparing to run routing scripts... Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=100 Jul 23 20:32:26 [2949] DBG:maxfwd:is_maxfwd_present: value = 70 Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=8 Jul 23 20:32:26 [2949] DBG:core:parse_to: end of header reached, state=10 Jul 23 20:32:26 [2949] DBG:core:parse_to: display={"551133239212"}, ruri={sip:551133239212@sps0.matrix.net.br} Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: <To> [53]; uri=[sip:551133239212@sps0.matrix.net.br] Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: to body ["551133239212"<sip:551133239212@sps0.matrix.net.br> ] Jul 23 20:32:26 [2949] DBG:siputils:has_totag: no totag Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=78 Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: cseq <CSeq>: <1> <INVITE> Jul 23 20:32:26 [2949] DBG:tm:t_lookup_request: start searching: hash=24133, isACK=0 Jul 23 20:32:26 [2949] DBG:tm:matching_3261: RFC3261 transaction matching failed Jul 23 20:32:26 [2949] DBG:tm:t_lookup_request: no transaction found Jul 23 20:32:26 [2949] DBG:core:parse_to_param: tag=b45cd830 Jul 23 20:32:26 [2949] DBG:core:parse_to: end of header reached, state=29 Jul 23 20:32:26 [2949] DBG:core:parse_to: display={"SPS0"}, ruri={sip:rafael@sps0.matrix.net.br} Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:26 [2949] DBG:core:grep_sock_info: no match for: [sps0.matrix.net.br:5060] Jul 23 20:32:26 [2949] DBG:core:grep_aliases: comparing host [0:sps0.matrix.net.br:5060] with us [2:sps0.matrix.net.br:5060] Jul 23 20:32:26 [2949] DBG:core:grep_aliases: match found for: [0:sps0.matrix.net.br:5060] Jul 23 20:32:26 [2949] DBG:core:check_self: host == me Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=10000 Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: content_length=361 Jul 23 20:32:26 [2949] DBG:core:get_hdr_field: found end of header Jul 23 20:32:26 [2949] DBG:auth:pre_auth: credentials with given realm not found Jul 23 20:32:26 [2949] DBG:auth:reserve_nonce_index: second= 16, sec_monit= -1, index= 2 Jul 23 20:32:26 [2949] DBG:auth:build_auth_hf: nonce index= 2 Jul 23 20:32:26 [2949] DBG:auth:build_auth_hf: 'Proxy-Authenticate: Digest realm="sps0.matrix.net.br", nonce="4a68f32800000002a304ed8610abc47da56bdc29957a394d" ' Jul 23 20:32:26 [2949] DBG:sl:send_reply: reply in stateless mode (sl) Jul 23 20:32:26 [2949] DBG:core:parse_headers: flags=ffffffffffffffff Jul 23 20:32:26 [2949] DBG:core:check_via_address: params 200.196.28.25, 192.168.5.38, 0 Jul 23 20:32:26 [2949] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:26 [2949] DBG:core:receive_msg: cleaning up Jul 23 20:32:26 [2947] DBG:core:parse_msg: SIP Request: Jul 23 20:32:26 [2947] DBG:core:parse_msg: method: <ACK> Jul 23 20:32:26 [2947] DBG:core:parse_msg: uri: <sip:551133239212@sps0.matrix.net.br> Jul 23 20:32:26 [2947] DBG:core:parse_msg: version: <SIP/2.0> Jul 23 20:32:26 [2947] DBG:core:parse_headers: flags=2 Jul 23 20:32:26 [2947] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-a957a2771c264219-1---d8754z->; state=6 Jul 23 20:32:26 [2947] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jul 23 20:32:26 [2947] DBG:core:parse_via: end of header reached, state=5 Jul 23 20:32:26 [2947] DBG:core:parse_headers: via found, flags=2 Jul 23 20:32:26 [2947] DBG:core:parse_headers: this is the first via Jul 23 20:32:26 [2947] DBG:core:receive_msg: After parse_msg... Jul 23 20:32:26 [2947] DBG:core:receive_msg: preparing to run routing scripts... Jul 23 20:32:26 [2947] DBG:core:parse_headers: flags=8 Jul 23 20:32:26 [2947] DBG:core:parse_to_param: tag=f8f2ab2c1295e90ed7dbb499b30f44b2.e355 Jul 23 20:32:26 [2947] DBG:core:parse_to: end of header reached, state=29 Jul 23 20:32:26 [2947] DBG:core:parse_to: display={"551133239212"}, ruri={sip:551133239212@sps0.matrix.net.br} Jul 23 20:32:26 [2947] DBG:core:get_hdr_field: <To> [95]; uri=[sip:551133239212@sps0.matrix.net.br] Jul 23 20:32:26 [2947] DBG:core:get_hdr_field: to body ["551133239212"<sip:551133239212@sps0.matrix.net.br>] Jul 23 20:32:26 [2947] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Jul 23 20:32:26 [2947] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:26 [2947] DBG:core:receive_msg: cleaning up Jul 23 20:32:27 [2946] DBG:core:parse_msg: SIP Request: Jul 23 20:32:27 [2946] DBG:core:parse_msg: method: <INVITE> Jul 23 20:32:27 [2946] DBG:core:parse_msg: uri: <sip:551133239212@sps0.matrix.net.br> Jul 23 20:32:27 [2946] DBG:core:parse_msg: version: <SIP/2.0> Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=2 Jul 23 20:32:27 [2946] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-267c082e8463a77a-1---d8754z->; state=6 Jul 23 20:32:27 [2946] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jul 23 20:32:27 [2946] DBG:core:parse_via: end of header reached, state=5 Jul 23 20:32:27 [2946] DBG:core:parse_headers: via found, flags=2 Jul 23 20:32:27 [2946] DBG:core:parse_headers: this is the first via Jul 23 20:32:27 [2946] DBG:core:receive_msg: After parse_msg... Jul 23 20:32:27 [2946] DBG:core:receive_msg: preparing to run routing scripts... Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=100 Jul 23 20:32:27 [2946] DBG:maxfwd:is_maxfwd_present: value = 70 Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=8 Jul 23 20:32:27 [2946] DBG:core:parse_to: end of header reached, state=10 Jul 23 20:32:27 [2946] DBG:core:parse_to: display={"551133239212"}, ruri={sip:551133239212@sps0.matrix.net.br} Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: <To> [53]; uri=[sip:551133239212@sps0.matrix.net.br] Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: to body ["551133239212"<sip:551133239212@sps0.matrix.net.br> ] Jul 23 20:32:27 [2946] DBG:siputils:has_totag: no totag Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=78 Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Jul 23 20:32:27 [2946] DBG:tm:t_lookup_request: start searching: hash=24130, isACK=0 Jul 23 20:32:27 [2946] DBG:tm:matching_3261: RFC3261 transaction matching failed Jul 23 20:32:27 [2946] DBG:tm:t_lookup_request: no transaction found Jul 23 20:32:27 [2946] DBG:core:parse_to_param: tag=b45cd830 Jul 23 20:32:27 [2946] DBG:core:parse_to: end of header reached, state=29 Jul 23 20:32:27 [2946] DBG:core:parse_to: display={"SPS0"}, ruri={sip:rafael@sps0.matrix.net.br} Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: no match for: [sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:grep_aliases: comparing host [0:sps0.matrix.net.br:5060] with us [2:sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:grep_aliases: match found for: [0:sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:check_self: host == me Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=10000 Jul 23 20:32:27 [2946] DBG:auth:check_nonce: comparing [4a68f32800000002a304ed8610abc47da56bdc29957a394d] and [4a68f32800000002a304ed8610abc47da56bdc29957a394d] Jul 23 20:32:27 [2946] DBG:core:db_new_result: allocate 28 bytes for result set at 0x814db68 Jul 23 20:32:27 [2946] DBG:db_mysql:db_mysql_get_columns: 1 columns returned from the query Jul 23 20:32:27 [2946] DBG:core:db_allocate_columns: allocate 4 bytes for result names at 0x8145e18 Jul 23 20:32:27 [2946] DBG:core:db_allocate_columns: allocate 4 bytes for result types at 0x8149f50 Jul 23 20:32:27 [2946] DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[0] at 0x8149798 Jul 23 20:32:27 [2946] DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x8149798)[0]=[password] Jul 23 20:32:27 [2946] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Jul 23 20:32:27 [2946] DBG:core:db_allocate_rows: allocate 8 bytes for rows at 0x8149c70 Jul 23 20:32:27 [2946] DBG:core:db_allocate_row: allocate 20 bytes for row values at 0x813e0c0 Jul 23 20:32:27 [2946] DBG:core:db_str2val: converting STRING [rmg25788] Jul 23 20:32:27 [2946] DBG:auth_db:get_ha1: HA1 string calculated: 25eb28681da92677e1e02a0e83d6f255 Jul 23 20:32:27 [2946] DBG:auth:check_response: our result = 'bd02426b09957f78a347036f020e4c48' Jul 23 20:32:27 [2946] DBG:auth:check_response: authorization is OK Jul 23 20:32:27 [2946] DBG:auth:post_auth: nonce index= 2 Jul 23 20:32:27 [2946] DBG:core:db_free_columns: freeing 1 columns Jul 23 20:32:27 [2946] DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x8149798 Jul 23 20:32:27 [2946] DBG:core:db_free_columns: freeing result names at 0x8145e18 Jul 23 20:32:27 [2946] DBG:core:db_free_columns: freeing result types at 0x8149f50 Jul 23 20:32:27 [2946] DBG:core:db_free_rows: freeing 1 rows Jul 23 20:32:27 [2946] DBG:core:db_free_row: freeing row values at 0x813e0c0 Jul 23 20:32:27 [2946] DBG:core:db_free_rows: freeing rows at 0x8149c70 Jul 23 20:32:27 [2946] DBG:core:db_free_result: freeing result set at 0x814db68 Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=14000 Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: content_length=361 Jul 23 20:32:27 [2946] DBG:core:get_hdr_field: found end of header Jul 23 20:32:27 [2946] DBG:core:comp_scriptvar: str 29 : rafael Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==9 && [sps0.matrix.net.br] == [127.0.0.1] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if host==us: 18==14 && [sps0.matrix.net.br] == [200.196.28.124] Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jul 23 20:32:27 [2946] DBG:core:grep_sock_info: no match for: [sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:grep_aliases: comparing host [0:sps0.matrix.net.br:5060] with us [2:sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:grep_aliases: match found for: [0:sps0.matrix.net.br:5060] Jul 23 20:32:27 [2946] DBG:core:check_self: host == me Jul 23 20:32:27 [2946] DBG:registrar:lookup: '551133239212' Not found in usrloc Jul 23 20:32:27 [2946] DBG:tm:t_newtran: transaction on entrance=(nil) Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=ffffffffffffffff Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=78 Jul 23 20:32:27 [2946] DBG:tm:t_lookup_request: start searching: hash=24130, isACK=0 Jul 23 20:32:27 [2946] DBG:tm:matching_3261: RFC3261 transaction matching failed Jul 23 20:32:27 [2946] DBG:tm:t_lookup_request: no transaction found Jul 23 20:32:27 [2946] DBG:tm:run_reqin_callbacks: trans=0xafc1d4b0, callback type 1, id 0 entered Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=78 Jul 23 20:32:27 [2946] DBG:core:parse_headers: flags=ffffffffffffffff Jul 23 20:32:27 [2946] DBG:core:check_via_address: params 200.196.28.25, 192.168.5.38, 0 Jul 23 20:32:27 [2946] DBG:core:_shm_resize: resize(0) called Jul 23 20:32:27 [2946] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Jul 23 20:32:27 [2946] DBG:tm:set_timer: relative timeout is 500000 Jul 23 20:32:27 [2946] DBG:tm:insert_timer_unsafe: [4]: 0xafc1d578 (17700000) Jul 23 20:32:27 [2946] DBG:tm:set_timer: relative timeout is 30 Jul 23 20:32:27 [2946] DBG:tm:insert_timer_unsafe: [0]: 0xafc1d594 (47) Jul 23 20:32:27 [2946] DBG:tm:_reply_light: reply sent out. buf=0x814ddf8: SIP/2.0 4..., shmem=0xafc1f148: SIP/2.0 4 Jul 23 20:32:27 [2946] DBG:tm:run_trans_callbacks: trans=0xafc1d4b0, callback type 256, id 0 entered Jul 23 20:32:27 [2946] DBG:tm:_reply_light: finished Jul 23 20:32:27 [2946] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jul 23 20:32:27 [2946] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:27 [2946] DBG:core:receive_msg: cleaning up Jul 23 20:32:27 [2948] DBG:core:parse_msg: SIP Request: Jul 23 20:32:27 [2948] DBG:core:parse_msg: method: <ACK> Jul 23 20:32:27 [2948] DBG:core:parse_msg: uri: <sip:551133239212@sps0.matrix.net.br> Jul 23 20:32:27 [2948] DBG:core:parse_msg: version: <SIP/2.0> Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=2 Jul 23 20:32:27 [2948] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-267c082e8463a77a-1---d8754z->; state=6 Jul 23 20:32:27 [2948] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jul 23 20:32:27 [2948] DBG:core:parse_via: end of header reached, state=5 Jul 23 20:32:27 [2948] DBG:core:parse_headers: via found, flags=2 Jul 23 20:32:27 [2948] DBG:core:parse_headers: this is the first via Jul 23 20:32:27 [2948] DBG:core:receive_msg: After parse_msg... Jul 23 20:32:27 [2948] DBG:core:receive_msg: preparing to run routing scripts... Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=8 Jul 23 20:32:27 [2948] DBG:core:parse_to_param: tag=bbd8f31c58bc6a3266ec5509a254d55b-2513 Jul 23 20:32:27 [2948] DBG:core:parse_to: end of header reached, state=29 Jul 23 20:32:27 [2948] DBG:core:parse_to: display={"551133239212"}, ruri={sip:551133239212@sps0.matrix.net.br} Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: <To> [95]; uri=[sip:551133239212@sps0.matrix.net.br] Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: to body ["551133239212"<sip:551133239212@sps0.matrix.net.br>] Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=100 Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: cseq <CSeq>: <2> <ACK> Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: content_length=0 Jul 23 20:32:27 [2948] DBG:core:get_hdr_field: found end of header Jul 23 20:32:27 [2948] DBG:maxfwd:is_maxfwd_present: max_forwards header not found! Jul 23 20:32:27 [2948] DBG:siputils:has_totag: totag found Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=200 Jul 23 20:32:27 [2948] DBG:rr:find_first_route: No Route headers found Jul 23 20:32:27 [2948] DBG:rr:loose_route: There is no Route HF Jul 23 20:32:27 [2948] DBG:core:parse_headers: flags=78 Jul 23 20:32:27 [2948] DBG:tm:t_lookup_request: start searching: hash=24130, isACK=1 Jul 23 20:32:27 [2948] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d8754z-267c082e8463a77a-1---d8754z- Jul 23 20:32:27 [2948] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Jul 23 20:32:27 [2948] DBG:tm:t_lookup_request: transaction found (T=0xafc1d4b0) Jul 23 20:32:27 [2948] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Jul 23 20:32:27 [2948] DBG:tm:insert_timer_unsafe: [2]: 0xafc1d4f8 (22) Jul 23 20:32:27 [2948] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jul 23 20:32:27 [2948] DBG:core:destroy_avp_list: destroying list (nil) Jul 23 20:32:27 [2948] DBG:core:receive_msg: cleaning up Jul 23 20:32:27 [2950] DBG:tm:utimer_routine: timer routine:4,tl=0xafc1d578 next=(nil), timeout=17700000 Jul 23 20:32:31 [2950] DBG:tm:timer_routine: timer routine:2,tl=0xafc1d4f8 next=(nil), timeout=22 Jul 23 20:32:31 [2950] DBG:tm:wait_handler: removing 0xafc1d4b0 from table Jul 23 20:32:31 [2950] DBG:tm:delete_cell: delete transaction 0xafc1d4b0 Jul 23 20:32:31 [2950] DBG:tm:wait_handler: done Jul 23 20:32:34 [2947] DBG:core:udp_rcv_loop: probing packet received from 200.202.22.5 55325 ------------------------------------------------------------- My Config File # # $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=7 log_stderror=yes log_facility=LOG_LOCAL7 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:mypassword@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://root:mypassword@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE) #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ------------------------------------------------------------- kamctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 3 | +1 | | 1 | 1 | | 4 | +55 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateways +---------+----------------+----------+------+------------+-----------+----- ---+-------+------+-------+--------+------+ | gw_name | ip_addr | hostname | port | uri_scheme | transport | grp_id | strip | tag | flags | weight | ping | +---------+----------------+----------+------+------------+-----------+----- ---+-------+------+-------+--------+------+ | MERAI | 66.165.166.236 | | 5061 | 1 | 2 | 1 | 0 | | 0 | 1 | 0 | | MERAII | 66.165.166.237 | | 5061 | 1 | 2 | 1 | 0 | | 0 | 1 | 0 | +---------+----------------+----------+------+------------+-----------+----- ---+-------+------+-------+--------+------+ ------------------------------------------------------------- kamctl dispatcher show dispatcher gateways +----+-------+-------------------------+-------+----------+-------------+ | id | setid | destination | flags | priority | description | +----+-------+-------------------------+-------+----------+-------------+ | 2 | 1 | sip:66.165.166.236:5061 | 1 | 1 | MERA | | 3 | 2 | sip:66.165.166.237:5061 | 3 | 0 | MERAII | +----+-------+-------------------------+-------+----------+-------------+ Rafael Moreno Matrix Telecom www.matrixtelecom.net.br / Phone: 55.11.3323-9207

1 0

[Serusers] incomplete INVITE-ACK-BYE pairs in ACC table
by Ilya Pichugin 23 Jul '09

23 Jul '09

Hi all In my acc table I see incomplete INVITE-ACK-BYE pairs for callids, in some calls there is no BYE. In what way it is possible to force generate BYE when user switch off his UAC, for example? Thanks a lot for advices. --------------- Ilya Pichugin i.pichugin(a)ptl.ru

2 2

[Serusers] check_to function in uri_db
by 静于 23 Jul '09

23 Jul '09

why there is no check_to function in uri_db in ser 2.0 version? what function can I use instead of check_to and check_from in ser.cfg? ___________________________________________________________ 好玩贺卡等你发，邮箱贺卡全新上线！ http://card.mail.cn.yahoo.com/

2 1

[Kamailio-Users] event_route[tm:local-request]
by Daniel-Constantin Mierla 23 Jul '09

23 Jul '09

Hello, latest master branch of sip-router includes support to execute an event route when a request is generated by tm module. Right now is a compile option: make EXTRA_DEFS="-DWITH_EVENT_LOCAL_REQUEST" cfg make all make install The functionality covers what was previously provided by local_route in Kamailio 1.5. In addition, parts of the core were updated to allow fine control of building the output buffer for requests. Testing and feedback is very much appreciated. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/index.php/sip-router-bootcamp/

1 0

[Serusers] Need help for Installing SER as Proxy
by Kai Dietrich 21 Jul '09

21 Jul '09

Maybe I am stupid, but since two weeks I will install SER with RTP Proxy as Proxy in our Network. The Situation. We have an new PBX System, with has direct SIP support. In the local network all works fine. But we want to add external Clients without VPN. For this reason I have installed a new server with two network Devices. One Device has an Public IP from the Internet, the other one a local IP. On the VoIP Device, wich I want to connect to our PBX via Internet I have adminstrate the internal PBX IP as Registra/Proxy und the Public IP as Ourgoing Proxy. With the constellation and many configurations I have found in the Internet the Register works and I can call an Internal telephone. But from the same telephone I can't call the VoIP device. And also it looks so, that the RTP Proxy dosnet work. The third problem is, that the PBX will send the UDP Packets direct to the Internet and not to the Proxy. I have also tested siproxd. With this programm I have an working constallation. The Problem is, that only one Client can be register. But I want much more. Can you help me, what I must configure that it works? Kind regards, Kai Dietrich

1 0

[Serusers] call limit (capacity) per gateway
by Ilya Pichugin 21 Jul '09

21 Jul '09

Hi all! How can I set call limit (capacity) per gateway? What module does provide this feature? Thanks a lot. --------------- Ilya Pichugin i.pichugin(a)ptl.ru

1 0

[Kamailio-Users] nonce_reuse protection issues
by Klaus Darilion 21 Jul '09

21 Jul '09

Hi! I really wonder if the nonce_reuse protection feature is useful and if anybody uses it without problems. One problem I have is with retransmission: e.g: ----INV1 ---> <---407------ ----ACK-----> ----INV2------> here happens a delay to the INVITE (e.g. jam in the access uplink, SIP proxy slow, ... whatever) which causes a retransmission of the INVITE ----INV3------> (retransmission of INV2) the proxy processes INV2, authenticates the user successful and forwards the requests then the proxy processes INV3, finds out that the nonce is reused and sends back 407 --> client gives up, but the request was also forwarded by the proxy :-( How do you handle such a scenario? Do you always create the transaction before authentication? One other thing I just found out is that reuse-check is done after successful authentication - shouldn't it be done the other way round? regards Klaus

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Re: [Kamailio-Users] received=IP added in Via
by Klaus Darilion 21 Jul '09

21 Jul '09

Hi! catalina oancea schrieb: > No, actually the value in the received parameter is not equal to the > IP in the Via. I looked in the code and this seems to be the intended > behavior. It would be nice to be able to disable it, but I don't think > there is one. Then it is standard conform. If the request is received from a different address then advertised, this parameter must be added. This is because SIP requires to send the response to the IP address from which the request was received. regards klaus > > Thanks anyway > Catalina > > > 2009/7/16 Klaus Darilion <klaus.mailinglists(a)pernau.at>: >> catalina oancea wrote: >>> Form what I found in >>> http://lists.kamailio.org/pipermail/devel/2008-August/015080.html, it >>> seems to me that it's a bug that kamailio adds received=IP parameter >>> although force_rport was not called. >> The bug was something else. >> >>>> Does anybody know how to disable the received=IP Via parameter that is >>>> added by default from kamailio? For the case I want to disable it in, >>>> I have no NAT handling. Also, I have: >>>> dns=no >>>> rev_dns=no >> >> In this case "received" will only be added if the advertised IP is different >> than the IP from which the request was received. (according to RFC). >> >> So, you do not use rport but kamailio adds the received parameter and the >> value in the received parameter is identical to the IP in the Via? >> >> regards >> klaus >>

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[Kamailio-Users] sanity_check question
by Dubravko Caric 21 Jul '09

21 Jul '09

Hi all, I have a question regarding sanity module (kamailio 1.5.1). Module documentation says "The function returns true if one of the checks failed", but as you can see in case 1 (request is valid) and case 2 (request is not valid because bad request uri) requests always enter function block. Does anybody else have the same problem? Any help would be appreciated... Thanks a lot /dubravko *Case 1:* if (is_method("REGISTER")) { xlog("L_INFO", "ENTER REGISTER\n"); if (sanity_check("128","6")) { xlog("L_NOTICE", "SANITY $si:$sp TIME $Tf USER $fu METHOD $rm\n"); exit; } } ENTER REGISTER Jul 20 13:30:13 [8180] DBG:core:parse_headers: flags=2000 Jul 20 13:30:13 [8180] DBG:core:get_hdr_field: cseq <CSeq>: <1> <REGISTER> Jul 20 13:30:13 [8180] DBG:core:get_hdr_field: content_length=0 Jul 20 13:30:13 [8180] DBG:core:parse_headers: flags=ffffffffffffffff Jul 20 13:30:13 [8180] DBG:core:get_hdr_field: found end of header Jul 20 13:30:13 [8180] DBG:sanity:sanity_check: all sanity checks passed SANITY xxx.xxx.xxx.xxx:22355 TIME Mon Jul 20 13:30:13 2009 USER sip:bob@exmaple.com METHOD REGISTER Jul 20 13:30:13 [8180] DBG:core:destroy_avp_list: destroying list (nil) Jul 20 13:30:13 [8180] DBG:core:receive_msg: cleaning up *Case 2:* ... if (sanity_check("130","6")) { ... ENTER REGISTER Jul 20 13:34:06 [8226] WARNING:sanity:check_ruri_scheme: failed to parse request uri Jul 20 13:34:06 [8226] DBG:core:parse_headers: flags=2000 Jul 20 13:34:06 [8226] DBG:core:get_hdr_field: cseq <CSeq>: <1> <REGISTER> Jul 20 13:34:06 [8226] DBG:core:get_hdr_field: content_length=0 Jul 20 13:34:06 [8226] DBG:core:parse_headers: flags=ffffffffffffffff Jul 20 13:34:06 [8226] DBG:core:get_hdr_field: found end of header Jul 20 13:34:06 [8226] SANITY xxx.xxx.xxx.xxx:22359 TIME Mon Jul 20 13:34:06 2009 USER sip:bob@exmaple.com METHOD REGISTER Jul 20 13:34:06 [8226] DBG:core:destroy_avp_list: destroying list (nil) Jul 20 13:34:06 [8226] DBG:core:receive_msg: cleaning up

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[Kamailio-Users] Runtime errors from libpurple when using purple module
by German Becker 20 Jul '09

20 Jul '09

Hi, I'm trying to test the purple module, and I am getting the following runtime errors, which I think are generated by libpurple: ** (process:13916): CRITICAL **: x509_tls_peers_init: assertion `ret == 0' failed ** (process:13916): CRITICAL **: purple_presence_set_status_active: assertion `status != NULL' failed ** (process:13916): CRITICAL **: purple_presence_set_status_active: assertion `status != NULL' failed ** (process:13916): CRITICAL **: purple_presence_is_online: assertion `presence != NULL' failed ** (process:13916): CRITICAL **: purple_status_type_get_attr: assertion `status_type != NULL' failed ** (process:13916): CRITICAL **: purple_status_type_get_id: assertion `status_type != NULL' failed ** (process:13916): CRITICAL **: purple_account_set_status_list: assertion `status_id != NULL' failed ** (process:13916): CRITICAL **: purple_status_type_get_attr: assertion `status_type != NULL' failed ** (process:13916): CRITICAL **: purple_status_type_get_id: assertion `status_type != NULL' failed ** (process:13916): CRITICAL **: purple_account_set_status_list: assertion `status_id != NULL' failed Attached the logfile. Any ideas about what might be causing this? I'm using version 1.5.0

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Re: [Kamailio-Users] 603 goes to onreply_route?
by catalina oancea 18 Jul '09

18 Jul '09

Even if I don't activate reply_route, it still doesn't use the failure route. I also tried 503 and 403, for these replies kamailio goes through both reply and failure routes. 2009/7/17 Klaus Darilion <klaus.mailinglists(a)pernau.at>: > > > catalina oancea schrieb: >> >> In my case, when 603 reply arrives, only the reply route is triggered. >> I added debug messages in both reply and failure routes and only the >> one in reply route is printed. > > What happens if you do not activated the reply_route. Is then the > failure_route called? > > Is it only for 603 or for all responses >400 ? > > regards > klaus > >> >> >> 2009/7/17 Klaus Darilion <klaus.mailinglists(a)pernau.at>: >>> >>> catalina oancea schrieb: >>>> >>>> Hello >>>> >>>> From my tests I found the following situation: >>>> >>>> Kamailio ------INVITE----------> I set t_on_failure and t_on_reply >>>> Kamailio <----100 Trying------- >>>> Kamailio <----603 Declined--- Here the reply goes in onreply_route (I >>>> added an xlog to prove it), not failure_route as expected. Is this the >>>> intended behavior or is it a bug (if it's a bug I can provide more >>>> details)? >>> >>> IIRC both routes should be executed. >>> >>> regards >>> klaus >>> >

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[Kamailio-Users] 603 goes to onreply_route?
by catalina oancea 17 Jul '09

17 Jul '09

Hello >From my tests I found the following situation: Kamailio ------INVITE----------> I set t_on_failure and t_on_reply Kamailio <----100 Trying------- Kamailio <----603 Declined--- Here the reply goes in onreply_route (I added an xlog to prove it), not failure_route as expected. Is this the intended behavior or is it a bug (if it's a bug I can provide more details)? Thanks, Catalina

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Re: [Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Henning Westerholt 17 Jul '09

17 Jul '09

On Donnerstag, 16. Juli 2009, Uriel Rozenbaum wrote: > I downloaded 1.5.2-notls and installed it. Hello Uriel, > The issue was fixed but I get the following: > Jul 16 16:56:52 sbci1pri /opt/kamailio/sbin/kamailio[3921]: > INFO:core:init_mod: initializing module exec > Jul 16 16:56:52 sbci1pri /opt/kamailio/sbin/kamailio[3921]: > ERROR:carrierroute:domain_fixup: could not find domain name '0' in map you need to specify the domain name, not the id. The id is used internally, the name is what is visible external. > [..] > In domain_name table i have this configured: > id domain > 0 domain0 > 1 domain1 Just try "domain0" as parameter. Cheers, Henning

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Re: [Kamailio-Users] received=IP added in Via
by catalina oancea 17 Jul '09

17 Jul '09

No, actually the value in the received parameter is not equal to the IP in the Via. I looked in the code and this seems to be the intended behavior. It would be nice to be able to disable it, but I don't think there is one. Thanks anyway Catalina > 2009/7/16 Klaus Darilion <klaus.mailinglists(a)pernau.at>: >> catalina oancea wrote: >>> >>> Form what I found in >>> http://lists.kamailio.org/pipermail/devel/2008-August/015080.html, it >>> seems to me that it's a bug that kamailio adds received=IP parameter >>> although force_rport was not called. >> >> The bug was something else. >> >>>> Does anybody know how to disable the received=IP Via parameter that is >>>> added by default from kamailio? For the case I want to disable it in, >>>> I have no NAT handling. Also, I have: >>>> dns=no >>>> rev_dns=no >> >> >> In this case "received" will only be added if the advertised IP is different >> than the IP from which the request was received. (according to RFC). >> >> So, you do not use rport but kamailio adds the received parameter and the >> value in the received parameter is identical to the IP in the Via? >> >> regards >> klaus >> >

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[Kamailio-Users] Kamailio Revision: 5487 1.4.3-notls (x86_64) -m 4096 -- Crashing Randomly
by Brandon Armstead 17 Jul '09

17 Jul '09

Hey guys, Here is the response from backtrace: Core was generated by `/sbin/kamailio -P /var/run/kamailio.pid -m 4096'. Program terminated with signal 11, Segmentation fault. [New process 29118] #0 0x00007f37499be44d in relay_reply (t=0x7f3645019f88, p_msg=0x857350, branch=0, msg_status=487, cancel_bitmap=0x7fff53aa3428) at t_reply.c:1101 1101 relayed_code=relayed_msg->REPLY_STATUS; (gdb) bt #0 0x00007f37499be44d in relay_reply (t=0x7f3645019f88, p_msg=0x857350, branch=0, msg_status=487, cancel_bitmap=0x7fff53aa3428) at t_reply.c:1101 #1 0x00007f37499bf8fb in reply_received (p_msg=0x857350) at t_reply.c:1404 #2 0x00000000004218fc in forward_reply (msg=0x857350) at forward.c:507 #3 0x000000000045cf35 in receive_msg ( buf=0x744aa0 "SIP/2.0 487 Request Terminated\r\nTo: <sip:+1XXXXXXXXXX@PSTN:5060>;tag=63c9393246a9a5d6i0\r\nFrom: \"XXXXX X & X \" <sip:XXXXXXXXXX@PSTN>;tag=VPSF506071629460\r\nCall-ID: STMMGC012009071521"..., len=997, rcv_info=0x7fff53aa3520) at receive.c:203 #4 0x000000000049c486 in udp_rcv_loop () at udp_server.c:449 #5 0x000000000042882b in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:780 As far as syslog/kernel message: Jul 15 21:07:28 sip-XXXX kernel: [526828.616918] kamailio[29118]: segfault at 40 ip 7f37499be44d sp 7fff53aa32b0 error 4 in tm.so[7f374997f000+4d000] The server load seems to randomly jump to ~ 8 - 10 and this happens, any thoughts / help is appreciated, thanks! One other thing that may be relevant, is we have disabled TCP support on this proxy, because previously (I have another mailing list thread) about, with TCP enabled kamailio will not start due to epoll alloc failure, thanks!

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[Serusers] Ser & RTPproxy does not change the SDP vodeo port.
by Amos Shahar 16 Jul '09

16 Jul '09

hi, I'm using ser 0.9.7 and RTPproxy 1.1 on diffrent servers. the ser.cfg is the nat-rtpproxy.cfg from the iptel site. I have 2 grandstream GXV-300 devices, each behind diffrent NAT, this setup is working fine for voice and RTP is relayed through the rtpproxy properly. The problem is that the video is not working, when ser gets the invite from device A and send it to device B, it change the audio port only but leave the video port the same as the port he got from device A. I'm not sure if it is an issue with the RTPproxy or the nat-helper module. any idea how to debug / resolve? do you aware of other versions of ser/rtpproxy that this setup should work with video? Thanks! Amos

1 0

[Kamailio-Users] received=IP added in Via
by catalina oancea 16 Jul '09

16 Jul '09

Hello, Does anybody know how to disable the received=IP Via parameter that is added by default from kamailio? For the case I want to disable it in, I have no NAT handling. Also, I have: dns=no rev_dns=no Anything else? Thanks Catalina

2 2

[Kamailio-Users] Clik to dial
by ALAEDDINE abbech 16 Jul '09

16 Jul '09

Hello every one I must use Openser alone to do the aplication clik to dial, who have an idea to do this application?? regards, Alasupcom

3 2

[Kamailio-Users] OpenSER stability problems in pilot project
by Munder Albert (CI/ISE) 16 Jul '09

16 Jul '09

Hello, Sorry, next try with posting this message. Attempts with attachment failed due to message size. We are running OpenSER in a pilot project and unfortunately have some stability problems. Any help or hints are appreciated. Project information OpenSER is used in a pilot project with * Appr. 5000 subscriber accounts * Appr. 1200 simultaneously registered users * Signalling encrypted with TLS * Media data encrypted with SRTP * Clients: softphones and hardphones * Re-registration time for clients: 3600 sec OpenSER configuration · Works as stateful SIP Proxy 1 mySQL database 2 Version 1.3.4.-TLS 3 Tcp_children: 100 --> is it recommended to increase this number? 4 Udp_children: 20 5 Tcp_connection_timeout: 3600 6 Shared memory: · -m 512 when error occurred 1 Now set to 1024 Problems * Shared memory consumption Shared memory usage is permanently increasing (about 50 MB per day) Application already crashed twice First messages were, these, repeated thousands of times (5915 times): Jun 17 08:54:52 si-.... /usr/local/sbin/openser[13921]: ERROR:core:tcpconn_new: shared memory allocation failure Jun 17 08:54:52 si-... /usr/local/sbin/openser[13921]: ERROR:core:handle_new_connect: tcpconn_new failed, closing socket And a few of these also (7613 times): Jun 17 08:57:24 si-... /usr/local/sbin/openser[13880]: ERROR:core:tls_accept: some error in SSL: Jun 17 08:57:24 si-... /usr/local/sbin/openser[13880]: ERROR:core:tls_print_errstack: error:1409C041:SSL routines:SSL3_SETUP_BUFFERS:malloc failure * TCP errors, lost SIP messages Examples from error messages: 14.100 times in log file from 17.06.09 Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:core:tcp_send: connect failed Jun 17 04:03:15 si-.. /usr/local/sbin/openser[13863]: ERROR:tm:msg_send: tcp_send failed Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:tm:t_forward_nonack: sending request failed Appears at least 20 000 times; and in the day of the last shared memory errors, it was 225.794 times in the log file (note that the number in parenthesis is usually 1 or 2, but on that day it has reached 6): Jun 17 09:01:27 si-.... /usr/local/sbin/openser[13921]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (6) Jun 17 09:01:27 si-... /usr/local/sbin/openser[13921]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (5) * Certificate validation problems TCP traffic is currently significantly increased by some ( appr. 70) clients which failed to validate the TLS certificate. Registration is repeated every 5 sec. Circa 30 thousand per day (on that day, it was 37.162 times in log) Jun 17 04:03:10 si-024lc008 /usr/local/sbin/openser[13801]: ERROR:core:tls_accept: some error in SSL: Jun 17 04:03:10 si-024lc008 /usr/local/sbin/openser[13801]: ERROR:core:tls_print_errstack: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca Mit besten Grüßen | Best regards Albert Munder Robert Bosch GmbH IT Systems Engineering (CI/ISE) Postfach 30 02 20 70442 Stuttgart GERMANY www.bosch.com Tel. +49 711 811-40562 Fax +49 711 811-5113333 Albert.Munder(a)de.bosch.com Robert Bosch GmbH, Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart HRB 14000 Aufsichtsratsvorsitzender: Hermann Scholl; Geschäftsführung: Franz Fehrenbach, Siegfried Dais; Bernd Bohr, Wolfgang Chur, Rudolf Colm, Gerhard Kümmel, Wolfgang Malchow, Peter Marks; Volkmar Denner, Peter Tyroller.

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[Kamailio-Users] no RTP randomly with rtpproxy
by Samuel Muller 16 Jul '09

16 Jul '09

A strange thing happens with RTP proxy since I compiled the v.1.2.0 release ... Radomly, (approx. 1 time per 10 calls) RTP proxy don't grab the caller's original source IP in the SDP ??? So I don't have any RTP, so no sound during the call. typical call schema : sip phone -> mitel ipbx -> kamailio -> audiocodes mediant 2000 -> pstn All the SIP headers and the SDP are ok each time ... the RTP ports are ok too, understood at each point. rtp proxy is launched with these options with a public IP address (and compiled with a modified port range) : /usr/sbin/rtpproxy -s udp:77.246.81.133 35000 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 77.246.81.133 -m 6000 -M 64000 kamailio is ok too in the config : kamailio IPv4 UDP sip.720.fr:46830-> rtpproxy.720.fr:35000 logs of RTP proxy (option -f to screen the debug) : # rtp ok : received command "29638_4 L 3962688592-61598521(a)10.33.146.4 77.246.81.136 35000 0_3962688592-61598523;1 1c1320846358;1" lookup on ports 6008/6010, session timer restarted pre-filling callee's address with 77.246.81.136:35000 sending reply "29638_4 6010 77.246.81.133 " caller's address filled in: 94.198.149.41:50186 (RTP) guessing RTCP port for caller to be 50187 # rtp not ok : received command "29571_2 L 3288148592-61598465(a)10.33.146.4 77.246.81.136 35000 0_3288148592-61598467;1 1c934295778;1" lookup on ports 6000/6002, session timer restarted pre-filling callee's address with 77.246.81.136:35000 sending reply "29571_2 6002 77.246.81.133 " and hop, no "caller's address filled in" ... it's exactly the same type a SIP call (with the same ip phone, etc...). I can precise that rtp proxy is running on the same machine that kamailio (v.1.4.4), but listen on a dedicated sub-if. Does someone knows something about that ? thanks ! .Samuel Muller. sml(a)720.fr

3 3

[Kamailio-Users] help for Multi-Domain
by zhangchao00001 16 Jul '09

16 Jul '09

Dear sir: I write to ask if the kamailio support multi-domain. for example, A UA belong to domain b get into domain a, could it work as well as in domain b? If so, how to configure it? Thank you.

4 3

[Kamailio-Users] sharing audio between SIP client and MSN client
by rachid 16 Jul '09

16 Jul '09

Hello all, I use kamailio to have sharing chat between a SIP client and MSN client Now i want to have a sharing audio between them but i don't know witch module to use?? someone can help me please -- ------------------------------------------------------------ BAHA Rachid Ingénieur Télécoms, INPT-Rabat GSM:(+33) 0619787609

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Re: [Kamailio-Users] softswitch
by peter brooks 16 Jul '09

16 Jul '09

i would like to offer people pc2pc phone service.can i do this with you software do you have a php admin page? can you install the software for a price? _________________________________________________________________ Windows Live™ SkyDrive™: Get 25 GB of free online storage. http://windowslive.com/online/skydrive?ocid=TXT_TAGLM_WL_SD_25GB_062009

2 1

[Serusers] Check out my photos on Facebook
by Tuhin Saiful 16 Jul '09

16 Jul '09

Hi serusers(a)iptel.org, I set up a Facebook profile where I can post my pictures, videos and events and I want to add you as a friend so you can see it. First, you need to join Facebook! Once you join, you can also create your own profile. Thanks, Saiful To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=1086097402&k=Z513ZV52PWZCUGAGYKYTWR&r 557896866 was invited to join Facebook by Tuhin Saiful. If you do not wish to receive this type of email from Facebook in the future, please click on the link below to unsubscribe. http://www.facebook.com/o.php?k=cebebd&u=557896866&mid=c8ac3aG2140d4a2G0G8 Facebook's offices are located at 1601 S. California Ave., Palo Alto, CA 94304.

1 0

[Kamailio-Users] question about $branch(name) PV
by Klaus Darilion 15 Jul '09

15 Jul '09

Hi! Does $branch(name) also handle the "main branch" ($ru, $du) as a branch, or will it handle only the additional branches? thanks klaus

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[Kamailio-Users] Carrierroute-Kamailio-1.4.3-Strip does not strip in the to: header
by Omar Mendoza 15 Jul '09

15 Jul '09

I have a prefix, and i remove using the Strip in the DB INVITE sip:8135553333@192.168.2.77 SIP/2.0 Record-Route: <sip:99998135553333@198.168.1.170;lr=on;ftag=as1074edd2> Via: SIP/2.0/UDP 198.168.1.170;branch=z9hG4bKa5b6.ce53.0 Via: SIP/2.0/UDP 198.168.1.180:5060;branch=z9hG4bK292133e9;rport=5060 From: "Omar" <sip:5125555555@198.168.1.180>;tag=as1074edd2 To: <sip:99998135553333@198.168.1.170> Contact: <sip:5125555555@198.168.1.180> __________________________ Omar

2 2

[Kamailio-Users] kamctl cr dump - Command not available on Kamailio 1.5.2
by Edson - Lists 15 Jul '09

15 Jul '09

Seems that an error is still there... Kamailio-151:~/sip-server # kamctl cr dump 500 command 'cr_dump' not available Kamailio-151:~/sip-server # kamailio -V version: kamailio 1.5.2-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, USE_SCTP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:5901M @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 13:29:16 Jul 14 2009 with gcc 4.3 Edson.

2 3

[Serusers] RTPproxy Fork Command
by Jeremy Valcourt 14 Jul '09

14 Jul '09

Hi, According to the RTPproxy 1.1 release notes you can fork a stream (http://www.rtpproxy.org/wiki/RTPproxy/RELENG_1_1_0_NOTES) : "The new C (copy) command has been introduced allowing to fork off one or both RTP streams in the session to some third-party in real time." However, after spending an afternoon searching the Internet, I can't for the life of me find any documentation on how to use this feature. So, how do you fork a stream with RTPproxy? Any help is greatly appreciated. Jeremy V.

1 0

[Kamailio-Users] OpenSer/Kamailio and A2Billing
by Carlos A. Alvarez 14 Jul '09

14 Jul '09

Folks, First of all I will like to apologize before hand, since I am better with numbers than with words. Therefore I hope you all can at least understand the essence of what I am trying to communicate. ;-) I am trying to set up an application, where I use Kamailio, asterisk, and A2Billing to process pre paid calling cards, and wholesale voip. I already have setup a kamailio service using carrier route and a2billing as a preprocessor. It works beautiful, Great work guys!!!! The questions I have are around the design for resiliency and redundancy. I will like to use another instance of kamailio for a dispatcher function to load balance between asterisk servers, since the is a limitation and performance issues with asterisk. Now I have done some research and some testing and they only way I know I can make it work is by using 302 redirection to redirect the invites from my customers to the corresponding asterisk server. If I use a direct setup using ds_select_dst asterisk gets all "confused" since the request appears to come from the dispatcher ip address. For wholesale ip, I filter a2billing on ip address and not username. My configuration is as follows: <customer>------><dispatcher/kamailio>------><a2billing/asterisk>-----><carrierroute/kamailio>------><carrier> Any thoughts, comments and/or suggestions are more than welcomed! Carlos

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[Kamailio-Users] Kamailio (OpenSER) v1.5.2 Released
by Daniel-Constantin Mierla 14 Jul '09

14 Jul '09

A new release in 1.5 series is out. Kamailio (OpenSER) 1.5.2 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0 or 1.5.1 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.5.2/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/1.5.2/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.2/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/1.5.2/ Modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ What is new in 1.5.x release series is summarized in the announcement of v1.5.0: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes Note: Kamailio is the name of former OpenSER project. First version under Kamailio name was 1.4.0. Older versions will continue to use OpenSER name. The source tree for current development version is hosted by SIP Router project, you can test it via: http://sip-router.org/wiki/migration/kamailio-3.0-config Many thanks to everybody that helped with this release! Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] reminder: kamailio 1.5.2 today
by Daniel-Constantin Mierla 14 Jul '09

14 Jul '09

Hello, anyone that still has to commit something on SVN branch 1.5 @sourceforge.net, please do it before 14:00GMT, afterwards will start releasing 1.5.2. Thanks, Daniel

1 1

[Kamailio-Users] aliases
by asd 14 Jul '09

14 Jul '09

Hello, I would like to assign multiple DIDs (not sure how it's called in Kamailio) for one of my sip user. This sip user is registering with 7000 as username, and I would like to assign 7001, 7002 numbers to this sip user. I have tried using dbaliases function, so I have added 7001 and 7002 as aliases for 7000. If I call 7002, then alias is retrieved fine from db, I have no problem with that. But 7000 is receiving this call as 7000 (so he see 7000 as a called number, instead of 7002). is there a way to send "called number 7002" to 7000? Kind regards, Dmitri

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[Kamailio-Users] [nat][registrar] setting max_expires dynamically
by Pascal Maugeri 14 Jul '09

14 Jul '09

Hi How can we set the expires value of the REGISTER OK sent back to the clients in a dynamic way ? We set already a value with modparam("registrar", "max_expires", 120) but if the UA has been detected to be behind a NAT we want to apply a different value. Regards, Pascal

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Re: [Kamailio-Users] preparing kamailio 1.5.2
by Daniel-Constantin Mierla 14 Jul '09

14 Jul '09

Hello, thanks for testing and reporting back. Please cc all the time to the list, so everybody interested gets updated about conclusions. Daniel On 14.07.2009 0:05 Uhr, Marco Bungalski wrote: > Hello, Daniel! > > Thank you for you documentation - with this my linux-knowledge increased .-) > > > I've just downloaded an testet - now Kamailio don't crash with a loaded > dialog-module after > a receive of SIP-message containing an empty record-route-field. So good > luck for tomorrow :) > > Message now: > Jul 13 23:59:37 [23474] ERROR:core:print_rr_body: failed to parse RR > Jul 13 23:59:37 [23474] ERROR:dialog:populate_leg_info: failed to print > route records > > > Thanks and cheers, > > Marco > > > -----Ursprüngliche Nachricht----- > Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Gesendet: Montag, 13. Juli 2009 17:05 > An: Marco Bungalski; kamailio > Betreff: Re: AW: AW: [Kamailio-Users] preparing kamailio 1.5.2 > > Hello Marco, > > On 13.07.2009 14:28 Uhr, Marco Bungalski wrote: > >> Hello, Daniel, >> >> i know how to compile, but where to download the version with the bugfix >> included? >> >> > if you fetch from svn the branch 1.5 then it has the fix. Here is a > short description of how the versioning was done so far for kamilio > (openser) from svn: > - when a major version is released, a branch with the first two numbers > in the version is created (e.g., 1.5) > - for each minor version (including the .0 - corresponding to the major > version) a tag is created with name of the three numbers in version > - any fix for the major version is committed in the branch. Once a new > minor release is out, a new tag is created. Therefore, the branch has > all the time the most updated code for a specific major release. The > tags are snapshots at the time of release. > > For the case of 1.5.x series: > - there is one branch 1.5 > - now there are two tags on svn: 1.5.0 and 1.5.1, soon there will be 1.5.2. > - the tarballs are created from a svn checkout done as exampled in the > wiki page I sent to you. > > probably reading a bit about svn (subversion) will help you. Anyhow, > next major version will use GIT. > > Now, just to download the latest 1.5.x version, do: > svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 > kamailio-1.5 > > You need subversion package installed. The continue installation as you > do with the tarball. > > Cheers, > Daniel > >> Hello, >> >> this one should help you: >> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn >> >> Cheers, >> Daniel >> >> On 07/13/2009 01:30 PM, Marco Bungalski wrote: >> >> >>> Hello, >>> >>> i am sorry to say that i've not testet it because i don't know where to >>> download the fix and how to implementate that :-( >>> >>> But if you give me a link i will test it today (i can simulate the bug) >>> >>> >> and >> >> >>> will send you a feedback. >>> >>> Greetings, >>> >>> Marco >>> >>> >>> >>> >>> Mit freundlichen Grüßen >>> >>> Marco Bungalski >>> >>> >>> Marco Bungalski GmbH >>> Traversale 5 >>> 27283 Verden >>> >>> Telefon: +49 4231 - 776 9999 >>> Fax: +49 4231 - 776 9998 >>> Mobil: +49 172 4204774 >>> >>> e-mail: Marco(a)Bungalski.de >>> web: www.bungalski.de >>> >>> Geschäftsführender Gesellschafter: Marco Bungalski >>> Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 >>> >>> -----Ursprüngliche Nachricht----- >>> Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Gesendet: Montag, 13. Juli 2009 12:32 >>> An: Marco Bungalski >>> Cc: users(a)lists.kamailio.org >>> Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 >>> >>> Hello >>> >>> On 07/13/2009 12:28 PM, Marco Bungalski wrote: >>> >>> >>> >>>> Hello, Daniel! >>>> >>>> >>>> >>>> Will the bug "Kamailio crashes when receiving an empty >>>> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >>>> >>>> >>>> >>>> If yes: Thanks for that :-) >>>> >>>> >>>> >>>> >>> it should be already in the svn (therefore yes). Have you tested it? >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >> >> > > >

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[Kamailio-Users] new module: mi_rpc
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, several days ago, a new module named mi_rpc was introduced to source tree. The goal is to be able to execute MI commands using RPC interface from sip router core. One big advantage of this is the executing MI commands using the sercmd tool. sercmd is a command line interface, able to do auto completion for commands, used to change parameters, get insights of core and modules at runtime. If you grabbed the source code (some guidelines at: http://sip-router.org/wiki/migration/kamailio-3.0-config), then: cd utils/sercmd make ./sercmd See the readme for more options. You need to compile and load modules_s/ctl: http://sip-router.org/docbook/sip-router/branch/master/modules_s/ctl/ctl.ht… Back to mi_rpc, one issue that needs to be sorted out is the output format. Right now doing a pretty-format printing which is not suitable for xmlrpc. Feedback is very much appreciated, thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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Re: [Kamailio-Users] How to know if a UAC is behind NAT after fix_nated_register() ?
by Pascal Maugeri 13 Jul '09

13 Jul '09

I answer to myself (it may help others) as I could test successfully the configuration below. Also there is a typo (isbflagset instead of isflagset): so when I detect a UA is behind NAT, I do: if (nat_uac_test("19")) { if (method=="REGISTER") { setbflag(7); fix_nated_register(); ... Then when my B2BUA sends invite to this UA I do (after lookup("location") ): lookup("location"); ... if (is_method("INVITE")) { if (isbflagset(7)) { setflag(5); Regards, Pascal On Mon, Jul 13, 2009 at 2:43 PM, Pascal Maugeri <pascal.maugeri(a)gmail.com>wrote: > I'm not that familiar with flag management. > > so when I detect a UA is behind NAT, I do: > > if (nat_uac_test("19")) { > if (method=="REGISTER") { > setbflag(7); > fix_nated_register(); > ... > > Then when my B2BUA sends invite to this UA I do (after lookup("location") > ): > > if (is_method("INVITE")) { > if (isflagset(7)) { > setflag(5); > > Is that correct ? > > BTW, regarding the flag number, is there a table of reserved numbers ? Can > I use "7" in this case ? > > -pascal > > > > > On Mon, Jul 13, 2009 at 12:00 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> On 07/13/2009 11:54 AM, Pascal Maugeri wrote: >> >>> Hi >>> >>> A quick question, when I apply a fix_nated_register() to a remote user >>> when he/she registers, how can I know later on if this user is behind NAT ? >>> >> set a branch flag for REGISTER before save location. After lookup location >> that flag is recovered and you can test it in the config file. >> >> Cheers, >> Daniel >> >> >>> All my calls go through a B2BUA: >>> >>> Caller ------> kamailio/rtpproxy ------> B2BUA ------> kamailio/rtpproxy >>> ------> Callee >>> >>> When my B2BUA calls a user, I want to know if this user is behind NAT or >>> not. For now all the calls go through rtpproxy but I would like to apply >>> this path only when user is behind NAT. >>> >>> Regards, >>> Pascal >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com/ >> >> >

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Re: [Kamailio-Users] preparing kamailio 1.5.2
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello Marco, On 13.07.2009 14:28 Uhr, Marco Bungalski wrote: > Hello, Daniel, > > i know how to compile, but where to download the version with the bugfix > included? > if you fetch from svn the branch 1.5 then it has the fix. Here is a short description of how the versioning was done so far for kamilio (openser) from svn: - when a major version is released, a branch with the first two numbers in the version is created (e.g., 1.5) - for each minor version (including the .0 - corresponding to the major version) a tag is created with name of the three numbers in version - any fix for the major version is committed in the branch. Once a new minor release is out, a new tag is created. Therefore, the branch has all the time the most updated code for a specific major release. The tags are snapshots at the time of release. For the case of 1.5.x series: - there is one branch 1.5 - now there are two tags on svn: 1.5.0 and 1.5.1, soon there will be 1.5.2. - the tarballs are created from a svn checkout done as exampled in the wiki page I sent to you. probably reading a bit about svn (subversion) will help you. Anyhow, next major version will use GIT. Now, just to download the latest 1.5.x version, do: svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio-1.5 You need subversion package installed. The continue installation as you do with the tarball. Cheers, Daniel > Hello, > > this one should help you: > http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn > > Cheers, > Daniel > > On 07/13/2009 01:30 PM, Marco Bungalski wrote: > >> Hello, >> >> i am sorry to say that i've not testet it because i don't know where to >> download the fix and how to implementate that :-( >> >> But if you give me a link i will test it today (i can simulate the bug) >> > and > >> will send you a feedback. >> >> Greetings, >> >> Marco >> >> >> >> >> Mit freundlichen Grüßen >> >> Marco Bungalski >> >> >> Marco Bungalski GmbH >> Traversale 5 >> 27283 Verden >> >> Telefon: +49 4231 - 776 9999 >> Fax: +49 4231 - 776 9998 >> Mobil: +49 172 4204774 >> >> e-mail: Marco(a)Bungalski.de >> web: www.bungalski.de >> >> Geschäftsführender Gesellschafter: Marco Bungalski >> Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 >> >> -----Ursprüngliche Nachricht----- >> Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Gesendet: Montag, 13. Juli 2009 12:32 >> An: Marco Bungalski >> Cc: users(a)lists.kamailio.org >> Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 >> >> Hello >> >> On 07/13/2009 12:28 PM, Marco Bungalski wrote: >> >> >>> Hello, Daniel! >>> >>> >>> >>> Will the bug "Kamailio crashes when receiving an empty >>> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >>> >>> >>> >>> If yes: Thanks for that :-) >>> >>> >>> >> it should be already in the svn (therefore yes). Have you tested it? >> >> Cheers, >> Daniel >> >> >> > >

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Re: [Kamailio-Users] preparing kamailio 1.5.2
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, this one should help you: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn Cheers, Daniel On 07/13/2009 01:30 PM, Marco Bungalski wrote: > Hello, > > i am sorry to say that i've not testet it because i don't know where to > download the fix and how to implementate that :-( > > But if you give me a link i will test it today (i can simulate the bug) and > will send you a feedback. > > Greetings, > > Marco > > > > > Mit freundlichen Grüßen > > Marco Bungalski > > > Marco Bungalski GmbH > Traversale 5 > 27283 Verden > > Telefon: +49 4231 - 776 9999 > Fax: +49 4231 - 776 9998 > Mobil: +49 172 4204774 > > e-mail: Marco(a)Bungalski.de > web: www.bungalski.de > > Geschäftsführender Gesellschafter: Marco Bungalski > Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 > > -----Ursprüngliche Nachricht----- > Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Gesendet: Montag, 13. Juli 2009 12:32 > An: Marco Bungalski > Cc: users(a)lists.kamailio.org > Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 > > Hello > > On 07/13/2009 12:28 PM, Marco Bungalski wrote: > >> Hello, Daniel! >> >> >> >> Will the bug "Kamailio crashes when receiving an empty >> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >> >> >> >> If yes: Thanks for that :-) >> >> > it should be already in the svn (therefore yes). Have you tested it? > > Cheers, > Daniel > > -- Daniel-Constantin Mierla http://www.asipto.com/

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Re: [Kamailio-Users] preparing kamailio 1.5.2
by Marco Bungalski 13 Jul '09

13 Jul '09

Hello, Daniel! Will the bug "Kamailio crashes when receiving an empty Record-Route-field with using the dialog-module" be fixed in 1.5.2? If yes: Thanks for that :-) Greetings, Marco

2 1

[Kamailio-Users] roadmap to kamailio 3.0
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, the roadmap to next major release of Kamailio, codenamed 3.0, was sketched during last week IRC meeting as: - freezing in 1-1.5 months - 1-1.5 months testing - release by end of September/ beginning of October Right now, current development state is: - all but one module (seas) were updated to work with the new core - several addons in kamailio core and tm still to be included in GIT master branch - many new features developed: mecached, event_route, xml doc handling ... see: http://sip-router.org/wiki/features/new-in-devel I invite all of you to test, startup guidelines at: http://sip-router.org/wiki/migration/kamailio-3.0-config Any feedback is very much appreciated, thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Kamailio-Users] How to know if a UAC is behind NAT after fix_nated_register() ?
by Pascal Maugeri 13 Jul '09

13 Jul '09

Hi A quick question, when I apply a fix_nated_register() to a remote user when he/she registers, how can I know later on if this user is behind NAT ? All my calls go through a B2BUA: Caller ------> kamailio/rtpproxy ------> B2BUA ------> kamailio/rtpproxy ------> Callee When my B2BUA calls a user, I want to know if this user is behind NAT or not. For now all the calls go through rtpproxy but I would like to apply this path only when user is behind NAT. Regards, Pascal

3 2

[Kamailio-Users] preparing kamailio 1.5.2
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, last week during the IRC meeting it was proposed to release kamailio 1.5.2 tomorrow. If anyone of you is aware of some show stopper, please reply. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Kamailio-Users] two more modules
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, two of the last three modules that were not ported to use the new sip router core are ready now: nat_traversal and siptrace. Particularly, siptrace was refurbished, should catch now forwarded ACK. Before release, there are couple of new features that should get in trunk since sip route core offers support for such extensions: - ability to filter the ACK for negative replies issued by sl module in the script - ability to store stateless forwarded messages (now only stateful ones being stored) Testing is very much appreciated. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Kamailio-Users] Kamailio 1.4.1 Configuration Issue
by dyerom AD 13 Jul '09

13 Jul '09

Good day everyone. As I am new to kamailio, I would like to ask help on the CFG file. I have successfully installed kamailio 1.4.1 but having problems to set it up on my desired functionality. Below is what i want to achieve. All sip clients (asterisk box, sip enabled box, pbx) should pass sip calls to my kamailio server. Clients connections to my Kamailio are via IP to IP. From my kamailio server i should be able to route them on my carrier, USA carrier and International Carrier. Connections to carrier is via IP to IP. I have set up Kamailio to support MYSQL authentication, its just that i have having problems to put my idea on the kamailio.cfg. Any sample working configurations similar to want I want to achieve is greatly appreciated. Thank you in advance for any support. Dennis Cristobal. jtom25(a)yahoo.com Feel safer online. Upgrade to the new, safer Internet Explorer 8 optimized for Yahoo! to put your mind at peace. It's free. Get IE8 here! http://downloads.yahoo.com/sg/internetexplorer/

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[Kamailio-Users] TCP supervisor process in Kamailio
by Pascal Maugeri 10 Jul '09

10 Jul '09

Hi I recently read the following in order to optimize OpenSER in handling TCP connections: "First, the TCP supervisor process must be given an elevated priority level in order to prevent anomalous behavior due to the Linux scheduler." First of all, as this is quite old paper (it refers to OpenSER 1.2), I'm wondering if such a tuning is still needed for Kamailio 1.5 branch ? If yes, how can I do this ? Regards, Pascal

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[Kamailio-Users] help for multi-domain
by zhangchao00001 09 Jul '09

09 Jul '09

Dear sir: I write to ask if the kamailio support multi-domain. for example, A UA belong to domain b get into domain a, could it work as well as in domain b? If so, how to configure it? Thank you.

2 1

force_rtp_proxy() vis-a-vis BYE
by Alex Balashov 08 Jul '09

08 Jul '09

According to the Kamailio 1.5.x docs, force_rtp_proxy() is deprecated and rtpproxy_offer() and rtpproxy_answer() should be used instead. Does that mean unforce_rtp_proxy() is also deprecated? If so, how should calls be cleaned up once a BYE is received? Is the orthodox approach to simply let the streams time out on their own, or should unforce_rtp_proxy() still be used here? What about when a CANCEL is received while the dialog is an early state -- should unforce_rtp_proxy() be used there too? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Serusers] SER generating Via: with branch parameter equal to "0" over half the time
by Frank Durda IV 08 Jul '09

08 Jul '09

A telco equipment vendor has pointed out that when an ACK message pass via our SER servers, a Via: header is added which is normal, but frequently the branch value in that Via: header is "0". As in: Via: SIP/2.0/UDP 10.133.90.x;branch=0 This violates RFC 3261 for being zero and for not being "unique across time and space". (8.1.1.7, page 38). This is showing up in over 50% of the ACK messages being passed through in just one direction. When not equal to zero, the value is like this: Via: SIP/2.0/UDP 10.133.90.x;branch=z9hG4bK095a.b27ac307.0 We don't see branch=0 appearing in any other SIP message type, just ACKs. Is anybody else seeing this and perhaps have a fix? The telco equipment vendor doesn't think it is confusing their equipment, but as it is out spec they want to eliminate it as a source of trouble. The calls are originating in a variety of types of equipment including Acme, Sonus, Asterisk, etc. This is the Via: added by SER, any already present when we got the call are non-zero and look fine. We are running ser-2.0.0-rc1. A sample: ACK sip:6x0x9x0x0x;npdi=yes;rn=6x0x6x9x7x@10.133.0.x:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1x3.9x.1x;branch=0 Via: SIP/2.0/UDP 7x.1x.1x3.1x0:5060;branch=z9hG4bK02Bd8a36ba5a45d76d6 From: <sip:Restricted@7x.1x.1x3.1x0>;tag=gK020a4ae7 To: <sip:6103900202@2x8.3x.2x1.7x>;tag=000a0285+1+11540020+75d5539b Call-ID: 2097284613_75974428(a)7x.1x.1x3.1x0 CSeq: 15498 ACK Max-Forwards: 16 Content-Length: 0 Thanks in advance.

2 3

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 07 Jul '09

07 Jul '09

Hello, within K project, we had from time to time an IRC meeting for short term planning. IMO it is the time for next one since probably we need at least a kamailio 1.5.2 release. The place was the irc channel #kamailio on freenode.net, now we can host it on #sip-router. Presence is not mandatory and anyone is welcome to join. From the perspective of K project, here are the topics I would like to approach: - kamailio 1.5.2 - show-stoppers, when? - shall we do another branch 1.4-based release - kamailio 3.0 - current status - what is missing, what can be worked-around, prioritizing tasks... - documentation - migration tutorials - release roadmap For sip router: - logistics: anything missing for project now? - what can be done to make adoption easier (new tools, etc...) - suggestions for future development I propose Tuesday, July 7, 2009, we can adjust to fit the best for the people that want to attend. I started a wiki page, feel free to add there: https://sip-router.org/wiki/devel/irc-meetings/2009-07-07 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

2 3

[Kamailio-Users] SEMS 1.1.1 released
by Stefan Sayer 07 Jul '09

07 Jul '09

(apologies if you receive multiple copies of this mail) Hello, the 111 version of SEMS, the SIP Express Media Server, is now available for download at http://ftp.iptel.org/pub/sems/1.1/1.1.1/ http://ftp.iptel.org/pub/sems/1.1/1.1.1/src/sems-1.1.1.tar.gz 4a6422d09ddadaf9eacd8cae8f0848d5 sems-1.1.1.tar.gz Debian packages for lenny64 are available in the repository: deb http://ftp.iptel.org/pub/sems/debian lenny free deb-src http://ftp.iptel.org/pub/sems/debian lenny free and etch64 in http://ftp.iptel.org/pub/sems/1.1/1.1.1/packages . This is a bugfix release in the 1.1 branch which accumulates fixes for bugs found in 1.1.0 so far. Specifically, this is - fixed Via HF missing the port number in ACK to 200 reply - do not try to scale too short RTP packets - fixed initialization of SSL - caused random crashing of xmlrpc server - fix size() for AmArg struct type - authenticate on both 401 and 407 reply in click2dial - fixed ssl build dependency for DIAMETER client in deb Many Thanks to everyone who contributed with bug reporting and fixes. More information, documentation etc about SEMS can be found at its homepage: http://iptel.org/sems Best Regards Stefan Sayer -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

1 0

[Kamailio-Users] force_rtp_proxy() vis-a-vis BYE
by Alex Balashov 07 Jul '09

07 Jul '09

According to the 1.5.x docs, force_rtp_proxy() is deprecated and rtpproxy_offer() and rtpproxy_answer() should be used instead. Does that mean unforce_rtp_proxy() is also deprecated? If so, how should calls be cleaned up once a BYE is received? Is the orthodox approach to simply let the streams time out on their own, or should unforce_rtp_proxy() still be used here? What about when a CANCEL is received while the dialog is an early state -- should unforce_rtp_proxy() be used there too? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

3 2

Images in the Document "SIP Introduction"
by Antônio Theóphilo 07 Jul '09

07 Jul '09

Hello Everybody Does anyone know how to get the images that are broken in the document "SIP Introduction" ( http://sip-router.org/docbook/sip-router/branch/master/sip/sip_introduction… )?It seems that they are diagrams very important for the understanding of the protocol's dynamics. Antônio Theóphilo

3 3

[Kamailio-Users] SIP-Router Project - how does this affect me?
by Geoffrey Mina 07 Jul '09

07 Jul '09

I am not clear on what exactly the SIP-Router project is, nor how it will affect me? Is Kamailio eventually being replaced with SIP-Router once all the functionalities have been rolled in? Will my entire configuration script need to be reworked to eventually operate with this new platform? Can anyone explain to me why this merge is happening so soon after Kamailio split with OpenSIPS? All this turmoil feels very destructive to me.

3 2

[Kamailio-Users] How should I change Calling Party Number for a call?
by Uriel Rozenbaum 07 Jul '09

07 Jul '09

Hi Guys, I was wondering how can I implement a proxy with Kamailio that can manage privacy and normalization of the calling party. In an INVITE message I can get the calling party name (or number) in several places: 1. From Header 2. Remote-Party ID 3. P-Asserted-Identity 4. P-Preferred-Identity How should I act in order to normalize or block the Calling Party to be passed to subsequent hops? AFAIK I can change the From Header as long as the tag isn't modified. Is this really true? What about RPID or PAI? Should I Change the aliases there as well? Thanks in advance to all! Uriel

3 2

[Kamailio-Users] strange fr_inv_timer behaviour
by hans＠onsmail.nl 07 Jul '09

07 Jul '09

Hi, When setting fr_inv_timer to 60s and the phone is not answered kamailio generates a timeout after 20s ? If fr_inv_timer is set to 90s the timeout is generated after 30s ? Doing a level 7 debug I received the following tm timer related output Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1755]: DBG:tm:set_timer: relative timeout is 500000 Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1755]: DBG:tm:insert_timer_unsafe: [4]: 0xafa6abc4 (16900000) Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1755]: DBG:tm:set_timer: relative timeout is 3 Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1755]: DBG:tm:insert_timer_unsafe: [0]: 0xafa6abe0 (51) Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1754]: DBG:tm:set_timer: relative timeout is 60 Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1754]: DBG:tm:insert_timer_unsafe: [1]: 0xafa6abe0 (108) Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1751]: DBG:tm:set_timer: relative timeout is 60 Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1751]: DBG:tm:insert_timer_unsafe: [1]: 0xafa6abe0 (108) Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:utimer_routine: timer routine:4,tl=0xafa6abc4 next=(nil), timeout=16900000 <phone is ringing ....> Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:timer_routine: timer routine:1,tl=0xafa6abe0 next=(nil), timeout=108 Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:final_response_handler: stop retr. and send CANCEL (0xafa6aa78) Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:t_should_relay_response: T_code=180, new_code=408 Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:t_pick_branch: picked branch 0, code 408 Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:t_check: start=0xafa6aa78 Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:t_check: transaction already found! Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:pv_get_tm_reply_code: reply code is <408> version: kamailio 1.5.1-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:5868M @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 14:40:00 Jun 3 2009 with gcc 4.1.2 Kind regards Hans

3 9

[Kamailio-Users] Polycom BLF presence
by Daniel Corbe 07 Jul '09

07 Jul '09

Anyone have any hints on how to make Polycom BLF presence working with Kamailio? -Daniel

3 2

[Kamailio-Users] RtpProxy: start_recording()
by Iñaki Baz Castillo 06 Jul '09

06 Jul '09

Hi, I'm using start_recording() function of nathelper module. I use "-R" option in rtpproxy daemon in order not to record RTCP. start_recording() is called in route and onreply_route. After the call ends I see the following files in the recording directory: 283K 2009-07-06 13:41 1e54fbae418504a40744ad765718792a(a)212.230.19.1=38a3b929-co1782-INS001;1.a.rtp 275K 2009-07-06 13:41 1e54fbae418504a40744ad765718792a(a)212.230.19.1=38a3b929-co1782-INS001;1.o.rtp 0 2009-07-06 13:41 1e54fbae418504a40744ad765718792a(a)212.230.19.1=38a3b929-co1782-INS001;2.a.rtp 0 2009-07-06 13:41 1e54fbae418504a40744ad765718792a(a)212.230.19.1=38a3b929-co1782-INS001;2.o.rtp Some questions: - "1.a.rtp" is one direction and "1.o.rtp" the other, right? - What are the last two files (0 size)? - How to extract the audio from these files? note that the used codec could be G729, ALAW, GSM... Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 10

Re: [SR-Users] [sr-dev] LCR/CR and prefix mask (REGEX?) [moved to SR-Users]
by Edson - Lists 06 Jul '09

06 Jul '09

I looked around... CR uses the same approach that LCR on prefix definitions (or, at least the README don't say/explain that it's different). So I went to look on REGEX module... Iñaki, this module has Your signature... let me ask You what exactly are this REGEX groups? Could I use something like: - Use RURI and recover associated "REGEX-group number"; - Ask LCR or CR to look for routes to prefix="REGEX-group number"; - proceed routing fail-over as usual... Edson Juha Heinanen escreveu: > Iñaki Baz Castillo writes: > > > I'd really like to see some mechanism to make routing decisions based > > on RURI username itself, rather than just no its prefix. Sure it > > could be a good feature. > > lcr module works on longest prefixes of r-uri username (= telephone > number). it was tried once operate on regular expressions, but it was > not a good idea. if you want something else than longest prefixes, > check if some other module (like drouting or cr) could provide that. > > -- juha > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >

3 2

Re: [Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Uriel Rozenbaum 06 Jul '09

06 Jul '09

Henning, how can I configure Kamailio to use some other path to find modules? i should make modules differently as well, right? I'll copy back the list so anyone can answer that. Rgds, Uriel On Wed, Jul 1, 2009 at 9:35 AM, Henning Westerholt < henning.westerholt(a)1und1.de> wrote: > On Mittwoch, 1. Juli 2009, you wrote: > > It's a CentOS 5.3 i386. > > > > # gcc --version > > gcc (GCC) 4.1.2 20071124 (Red Hat 4.1.2-42) > > Copyright (C) 2006 Free Software Foundation, Inc. > > This is free software; see the source for copying conditions. There is NO > > warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR > PURPOSE. > > > > # uname -a > > Linux sbci1pri.cpsnet.com.ar 2.6.18-92.1.18.el5 #1 SMP Wed Nov 12 > 09:30:27 > > EST 2008 i686 i686 i386 GNU/Linux > > > Hi Uriel, > > > thanks, but this is all fairly common, so i doubt your environment is > guilty here. > > > > Do you have some guidelines to erase all Kamailio files from the system? > > I'm sure its a build problem. > > > Sorry, this depends where you installed the server, you basically need to > track all files that were installed. I missed the "make uninstall" also a > few times. :/ > > > But you could just try to install the new version in another directory, > like /tmp/kamailio or something like this. If you then configure the server > to only use modules from this dir it should be not conflict with the old > installation. > > > Henning >

4 9

[Serusers] rtpproxy: How to force it to send packets out a specific interface?
by Frank Durda IV 06 Jul '09

06 Jul '09

I am running SER/rtpproxy on some systems with multiple cores and multiple ethernet interfaces. I am running multiple copies of rtpproxy, each with an IP address associated with a particular interface, as in one rtpproxy is using em1 which is 208.33.44.3 and em2 is 208.33.44.4 which another rtpproxy listens at, while ser listens in em0 at 208.33.44.2. These three (actually ten) interfaces pass through ethernet switches and then on to a router with a default gateway of 208.33.44.1 and then off to the Internet or wherever. (In my setup, each rtpproxy has a second interface pointing into a private network, so only one side of a call appears on any given interface and five face toward the Internet.) SER+rtpproxy correctly divides RTP traffic for calls to 208.33.44.3 and 208.33.44.4 in the SDP payloads, and callers send their RTP packets to the two rtproxies IP addresses without problems. That part works just fine. The problem is that rtpproxy is using the default route for transmission, so everything that goes towards the Internet goes out via em0, so em1 and em2 don't transmit anything. This means that this one link is overwhelmed by the traffic that should have been divided across multiple links. Rtpproxy is allowing the local route table and the default route to handle the outgoing packets and so the packets are all going out the same interface. In my case, em0 was 100Mbit/sec (completely sufficient for SIP messages, while em1 and em2 were gigabit links to handle RTP, but they are getting used only in one direction. Even if em0 was upgraded to Gbit, it would still not be able to handle the potential for 2+ Gbit of data from the rtproxies. So, is there any way to force rtproxy to perform what I believe they used to call "reflector mode", where the answers go out the same interface they come in on? (This used to be common on UDP-based NFS file servers.) A number of communication programs also allow this behavior as an option, but rtpproxy doesn't appear to do it or it isn't turned on by default. Because the destinations for the RTP are out on the Internet somewhere, I don't believe I can solve this via "route" or "ifconfig" settings, but would be delighted if someone knows different or knows of a setting change in rtpproxy that will do this so that load balancing will work. The OS in question is FreeBSD 7.2, but I have the same issue on 6.2 and 6.3. Thanks in advance!

3 3

Re: [Kamailio-Users] SIP-Router Project - how does this affect me?
by Alex Balashov 06 Jul '09

06 Jul '09

That said, this page does speak to a number of the concerns, even if somewhat vaguely: http://sip-router.org/answers/ > >> I am not clear on what exactly the SIP-Router project is, nor how it >> will affect me? Is Kamailio eventually being replaced with SIP-Router >> once all the functionalities have been rolled in? Will my entire >> configuration script need to be reworked to eventually operate with >> this new platform? Can anyone explain to me why this merge is >> happening so soon after Kamailio split with OpenSIPS? All this >> turmoil feels very destructive to me. > > This is a confusing topic, and could definitely use additional discussion > and official clarification. > > I have been made to understand that Kamailio will not be replaced by > sip-router. It is rather that sip-router is a a "common core" that > integrates both Kamailio and sip-router modules, and that both projects > will share this common core going forward, but remain distinct. The > analogy provided to me by Daniel was of Linux kernels vs. distributions. > > This seems fair enough, but it doesn't adequately explain who sip-router > is intended for and who Kamailio is intended for, and who should use which > and why and under what circumstances. It does also seem that most of the > development and documentation energy is being concentrated into sip-router > integration, which leaves open the question of what the future of Kamailio > is. > > This definitely needs to be discussed more. Thank you very much for > raising it, Geoffrey. > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (678) 237-1775 > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

2 1

[Kamailio-Users] Branch Flag / Location Lookup
by Brandon Armstead 04 Jul '09

04 Jul '09

Hey guys, First off thanks for any help in advanced! I'm saving a nat branch flag into usrloc when the user registers, now each user has two registrations to the same username in usrloc, upon lookup("location"), inside of branch_route[] it appears only one of them have the nat branch flag set, while the other does not? Is there any case where a branch flag could somehow get mangled, or what to look for, or what may cause this? Also one other thing from a database outlook (the cflag) is equivalent on both AOR's. However again at the time of branch_route, one flag is i.e. 000800, and the other 000820, any thoughts are appreciated, thanks!

4 6

Re: [Kamailio-Users] [LCR] About ping
by Iñaki Baz Castillo 04 Jul '09

04 Jul '09

2009/3/23 Alexandr Dubovikov <shurik(a)start4.info>: > On Mon, Mar 23, 2009 at 11:15:52AM +0100, Iñaki Baz Castillo wrote: >> 2009/4/1 Alexandr Dubovikov <shurik(a)start4.info>: >> > On Fri, Mar 20, 2009 at 10:21:20PM +0100, Andreas Heise wrote: >> >> Hello Iñaki, > > Hi Inaki, > >> > >> > Hi all, >> >> >> >> you should ask Alexandr he has introdused this feature with rev5452, >> >> but I'm not sure if he is on the lists all the time, so I'll forward your >> >> question >> >> to him... >> > >> > sorry, I am currently offline and couldn't answer directly to the list. >> > >> > anyway. I agreed, 180 seconds it's too big ping interval, but it couldn't be >> > also less 32 seconds. >> > >> > >> > http://www.ietf.org/rfc/rfc3261.txt >> > >> > 17.2.2 Non-INVITE Server Transaction >> > >> > >> > T1 = 500 ms. >> > >> > >> > Timer J 64*T1 for UDP Section 17.2.2 Wait time for >> > 0s for TCP/SCTP non-INVITE request >> > >> > so, the timer for "completed" is 32 seconds and couldn't be less. >> > >> > >> > of course, you can use less value, but on own risk :-) >> > RFC 3261 are not recommendet it :) >> >> Thanks for pointing it out. >> However, if a gw takes so long time (~32 seconds) to respond I prefer >> to consider it as "offline" :) > > don't forget to disable ping for this gateway, because finaly you will have > a DDOS effect of the "offline" gw. :-) Well, If the host is already offline then it doesn't matter to kill it more XD Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 6

[Kamailio-Users] Problem in adding new user in database
by ajay_ei 03 Jul '09

03 Jul '09

Hi, I am a new user of openser. I have installed and configured openser on my local system. when i am trying to add a new user in database i am getting following error: [root@Ajay-PC sbin]# openserctl add ajay ajay ajay@localhost database engine 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter=0 ERROR 1054 (42S22) at line 1: Unknown column 'phplib_id' in 'field list' ERROR: introducing the new user 'ajay' to the database failed Kindly help me. Thanking you. -- View this message in context: http://www.nabble.com/Problem-in-adding-new-user-in-database-tp24307793p243… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 5

[Kamailio-Users] incomplet uri
by towhid islam 02 Jul '09

02 Jul '09

why this is happning ? uuid | username | domain | attribute | value | type | modified | +------+----------+--------------------+-----------+----------------+------+---------------------+ | 1001 | 100001 | sip:192.168.30.131 | abc | 192.168.30.131 | 0 | 2009-07-02 07:36:05 | | 1002 | 100002 | sip:192.168.30.131 | abc | 192.168.30.131 | 0 | 2009-07-02 07:36:18 | | 1003 | 100003 | sip:192.168.30.131 | abc | 192.168.30.131 | 0 | 2009-07-02 07:36:29 | if(avp_db_load("$ruri/domain","$avp(s:abc)")){ xlog("L_INFO","testing"); } ERROR:avpops:ops_dbload_avps: incomplet uri <sip:192.168.30.131> TOWHID

3 2

[Kamailio-Users] forking on URIs from DB
by Robert McGilvray 02 Jul '09

02 Jul '09

I'm trying to setup custom aliases out of a sql db that will sometimes need to fork and need some help realizing how I can assign q values and process them correctly. I'm using kam 1.5. Let's assume I query for our main support line and return 3 URIs from the db along with a separate column for the q. Sip:us-support@mycomp.com q=1.0 Sip:uk-support@mycomp.com q=1.0 Sip:in-support@mycomp.com q=0.90 Based on the values I'd want to parallel fork on the first two to send this to my UK and US support line, if that fails cancel the first two branches and route to India. Just to see if I was on the right track I tried adding them as branches using $br, which from what I understand is the only way to add branches from avps/variables. I added them by appending ;q=1.0 on the end of the URI like you would see in a Contact header but I don't think that's valid. Calls to $(branch(q) return <null> so it obviously doesn't take. If I were to just fork sequentially I could just load them into avps and call it in a failure route, but how do I handle a combination of the two? Am I wrong in trying to specify the q along with the URI? I noticed serialize_branches was removed in 1.4, which is what I use in production today in 1.3 - do I need to use the replacements in TM to order them and add the branches? Thanks in advance. -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

3 6

[Kamailio-Users] Nathelp module Option message Empty
by Matteo Piazza 01 Jul '09

01 Jul '09

Hi to everybody, I have an issue with the nathelper module and openser 1.3.4 . Basically I want generate the Option message after the registration of a client, but the server generate a UDP message with the correct ip and port but empty. Below the openser.cfg file and as attachment a pcap file where you can see the registration message and the UDP message empty. Any clue? With the same configuration in another enviroment I don't have any issue Kind Regards Matteo # # $Id: openser.cfg 4424 2008-06-27 10:25:22Z henningw $ # # OpenSER basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "//etc/openser/tls/user/user-cert.pem" #tls_private_key = "//etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "//etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:77.72.192.99:5060 alias= leo.trentinonetowrk.it ####### Modules Section ######## #set module path mpath="//lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "domain.so" #loadmodule "presence.so" #loadmodule "presence_xml.so" loadmodule "dialog.so" loadmodule "lcr.so" loadmodule "avpops.so" loadmodule "pdt.so" loadmodule "permissions.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) # ---- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "max_contacts", 10) modparam("registrar", "received_avp", "$avp(s:rcv)") # ----- uri_db params ----- modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "postgres://openser:openserrw@localhost/openser") # ----- acc params ----- modparam("acc", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("acc", "early_media", 1) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) modparam("acc", "failed_transaction_flag", 3) #modparam("acc", "log_flag", 1) #modparam("acc", "log_missed_flag", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; to_user=$tU; source_ip=$si") # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("usrloc", "nat_bflag", 6) # ----- auth_db params ----- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- modparam("alias_db", "db_url", "postgres://openser:openserrw@localhost/openser") # ----- domain params ----- modparam("domain", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") # ----- avpops param ----- modparam("avpops","avp_url","postgres://openser:openserrw@localhost/openser") modparam("avpops","avp_table","avptable") modparam("avpops","avp_table","usr_preferences") modparam("avpops","use_domain",1) modparam("avpops","uuid_column","uuid") modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","attribute_column","attribute") modparam("avpops","value_column","value") modparam("avpops","type_column","type") # ----- lcr params ----- modparam("lcr","db_url","postgres://openser:openserrw@localhost/openser") modparam("lcr","fr_inv_timer",90) modparam("lcr","fr_inv_timer_next",30) modparam("lcr", "dm_flag", 25) modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "contact_avp", "$avp(i:711)") # ------- PDT ---------- modparam("pdt", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("pdt", "db_table", "pdt") modparam("pdt", "domain_column", "source_domain") modparam("pdt", "prefix_column", "prefix") modparam("pdt", "domain_column", "domain") #modparam("pdt","check_domain",0) #modparam("pdt", "char_list", "0123456789*+[]-.") # ---------- permission ------------- modparam("permissions", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "allow_suffix", ".allow") modparam("permissions", "deny_suffix", ".deny") # ---------- dialog ------------- modparam("dialog", "dlg_flag", 4) modparam("dialog", "dlg_match_mode", 1) # ------- nathelper ------------ modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7890") modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 0) #modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:ping@leo.trentinonetwork.it") route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); setbflag(7); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("1"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[1] { route(10); route(1); } -- ========================================================================== Matteo Piazza Trentino Network s.r.l. Area Innovazione Via Gilli, 2 - 38100 TRENTO Tel (+39) 0461.020224 Mob (+39) 335.5378482 Fax (+39) 0461.020201 Cap. Soc. sottoscritto 7.573.248,00 - i. v. REG. IMP. C.F. e P. IVA 01904880224 Società soggetta a direzione e controllo da parte della Provincia Autonoma di Trento. C.F. e P. IVA 00337460224 ==========================================================================

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[Kamailio-Users] kamailio with sqlops fails to start if DB is down
by logivoip＠gmail.com 01 Jul '09

01 Jul '09

Hi All, First off - thank you for this module. I have configured two redundant DB sources in my kamailio confiuration. The DB's are accessed from Kamailio for redundancy pruposes. Kamailio and the primary DB are on the same hardware node. The secondary DB is on a remote host. If Kamailio is up and both MySQL DB's are up, the config works like a charm. If the primary or the secondary DB becomes unreachable while Kamailio is up, the config handles the exception and all works well. However, if any of the DB servers becomes unreachable and Kamailio is restarted, Kamailio can no longer start. Below is an excerpt from my config file: # -- sqlops params modparam("sqlops","sqlcon","pridb=>mysql://root:@localhost/db_table") modparam("sqlops","sqlcon","secdb=>mysql://root:@192.168.1.231/db_table") /var/log/messages Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:db_mysql:db_mysql_new_connection: driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:core:db_do_init: could not add connection to the pool Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:sqlops:sql_connect: failed to connect to the database [pridb] Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:core:init_mod_child: failed to initializing module sqlops, rank 1 Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:core:main_loop: init_child failed for UDP listener What is the proper way of using multiple database servers with Kamailio? Thanks in advance. --JR

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[Kamailio-Users] kamailio at linuxtag 2009
by Henning Westerholt 01 Jul '09

01 Jul '09

Hi all, i'm pleased to announce that i give a talk about the kamailio project on the linuxtag 2009 in Berlin. The title of the talk is "Building carrier grade voice over IP systems with kamailio", on wednesday the 24th june afternoon. Further details can be found in the conference program at: http://www.linuxtag.org/2009/de/program/freies-vortragsprogramm/mittwoch/vo… I'll be also present on the linuxtag on thursday, so if you would like to meet for some discussions, just drop me a mail (off-list). With best regards, Henning Westerholt

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[Kamailio-Users] new feature: execute mi commands via rpc
by Daniel-Constantin Mierla 30 Jun '09

30 Jun '09

Hello, there is a new module in repository that allow you to execute most of the mi commands from Kamailio modules (see readme) via RPC interface inherited from SER: http://sip-router.org/docbook/sip-router/branch/master/modules/mi_rpc/mi_rp… http://sip-router.org/docbook/sip-router/branch/master/rpc/ser_rpc.html One of the nicest things you can out of this is usage of sercmd - command line interface tool. It is located in utils/sercmd. Thanks to Andrei, sercmd does now tab completion for mi command. How to use it: - get sip router from git: git clone --depth 1 git://git.sip-router.org/sip-router cd sip-router make all - add to your config file: loadmodule "ctl.so" # located in 'modules_s' loadmodule "mi_rpc.so" # located in 'modules' - use sercmd cd utils/sercmd make ./sercmd Then you should get: sercmd> mi <tab><tab> arg nh_show_rtpp t_reply ul_rm_contact cleanup ps t_uac_cancel ul_show_contact debug pwd t_uac_dlg uptime dum refreshWatchers ul_add version kill shv_get ul_dump which nh_enable_ping shv_set ul_flush nh_enable_rtpp t_hash ul_rm sercmd> mi uptime 200 OK + Now:: Tue Jun 30 12:47:54 2009 + Up since:: Tue Jun 30 12:47:26 2009 + Up time:: 28 [sec] Via sercmd you can control lot of internals at runtime -- see 'help' command -- more about: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ut… Enjoy, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] [ZIP-Attachment has been removed] OpenSER stability problems in pilot project
by Munder Albert (CI/ISE) 30 Jun '09

30 Jun '09

The ZIP-attachment sent to you was blocked. Please inform the original sender to rename ZIP-files. Die Ihnen gesandte ZIP-Datei wurde geblockt. Bitte informieren Sie den ursprünglichen Absender. (Dateiendung umbenennen). Sender: Albert.Munder(a)de.bosch.com Recipients: users(a)lists.kamailio.org Subject: "OpenSER stability problems in pilot project" Datei / Filename: OpenSER_stability.zip Time: Jun 30, 2009 11:49:04 AM The cleaned message is attached. ------------------------------------------------ Hello, We are running OpenSER in a pilot project and unfortunately have some stability problems. Related information are summarized in the attached document: Any help or hints are appreciated. Mit besten Grüßen | Best regards Albert Munder Robert Bosch GmbH IT Systems Engineering (CI/ISE) Postfach 30 02 20 70442 Stuttgart GERMANY www.bosch.com Tel. +49 711 811-40562 Fax +49 711 811-5113333 Albert.Munder(a)de.bosch.com Robert Bosch GmbH, Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart HRB 14000 Aufsichtsratsvorsitzender: Hermann Scholl; Geschäftsführung: Franz Fehrenbach, Siegfried Dais; Bernd Bohr, Wolfgang Chur, Rudolf Colm, Gerhard Kümmel, Wolfgang Malchow, Peter Marks; Volkmar Denner, Peter Tyroller.

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Re: [Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Uriel Rozenbaum 30 Jun '09

30 Jun '09

I'll try first with the debug in kamailio... I'll let you know in a few minutes... 2009/6/27 Raúl Alexis Betancor Santana <rabs(a)dimension-virtual.com> > On Saturday 27 June 2009 02:10:36 Uriel Rozenbaum wrote: > > Oki doki... how can I do that? I really don't know how to do it. > > I don't know witch server are you using, but on PostgreSQL, you have to set > an > option on the postgresql.conf file that allow your to dump by syslog all > the > DDL statements the server receives. > > I also beleive that kamailio, if you put it on debug level 6 or more, will > dump the SQL statements it's sending out. > > > -- > Raúl Alexis Betancor Santana > Dimensión Virtual >

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[Kamailio-Users] Kamailio / OpenSer billing platform
by Meftah Tayeb 30 Jun '09

30 Jun '09

hello all, please any billing modules or billing application to use with Kamailio / OpenSer? i need a prepay / Postpay application sam to Asterisk to billing (A2Billing) thanks for your help __________ Information from ESET NOD32 Antivirus, version of virus signature database 4191 (20090626) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

3 2

[Kamailio-Users] new feature: fine access to xml documents
by Daniel-Constantin Mierla 29 Jun '09

29 Jun '09

Hello, several days ago I committed support for accessing components of xml documents using pseudo-variables based on XPath syntax. The new $xml(...) PV is implemented in the presence_xml module as it works with xml docs and links libxml2. Short description in the cookbook: https://sip-router.org/wiki/cookbooks/pseudo-variables/devel#presence_xml_p… As an example: - publish has the body: <?xml version='1.0' encoding='UTF-8'?> <presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:daniel@asipto.com'> <tuple id='t8b484917'> <status><basic>open</basic></status> </tuple> <dm:person id='p7562ef60'><rpid:activities><rpid:unknown/></rpid:activities></dm:person> </presence> Getting the presence status: # this is needed because libxml2 cannot work with default no-prefix ns, so we alias the "p" prefix. modparam("presence_xml", "xml_ns", "p=urn:ietf:params:xml:ns:pidf") $xml(a=>doc) = $rb; $xml(a=>xpath:/p:presence/p:tuple/p:status/p:basic) Next is to update the presence modules to be able to work with the documents that have been updated in the config. Comments, opinions and improvements are welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Serusers] ser avpops {Scanned}
by sszasz＠digicomm.ro 29 Jun '09

29 Jun '09

Hi, I'm using SER and OpenSIPS for tests and I have a problem with avpops. How can I translate the following code snipplet ( working in opensips) to work in SER? if(avp_db_load("$fu/username","$avp(s:CBlocal)")) { sl_send_reply("403", "Call is barred"); exit; }; Thanks in advance Szasz Szabolcs -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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[Kamailio-Users] is avp the best option?
by Juan Perez 29 Jun '09

29 Jun '09

hello I need to load into memory a list of ip address associated with an index: ex: index, IP 1, 192.168.1.1 2. 192.168.1.2 3, 192.168.1.3 ......... ......... n, 192.168.1.n then at some point in the config I will call the variable, passing the index as argument and retrieving the IP as result can I do it with the avp module? cheers jp

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Re: [Kamailio-Users] sipsak
by Saúl Ibarra 29 Jun '09

29 Jun '09

Well, I think you should just need to locate the user and relay the request... but first you need to figure out why you afecta replying 501 to a notify... El 29 de jun de 2009, 11:32 a.m., "Chandrakant Solanki" < solanki.chandrakant(a)gmail.com> escribió: How can I explicitly permitting notify request from kamailio config...??? On Mon, Jun 29, 2009 at 2:45 PM, Saúl Ibarra <saghul(a)gmail.com> wrote: > > Are you explicitly not ... -- Regards, Chandrakant Solanki

1 0

[Kamailio-Users] sipsak
by Chandrakant Solanki 29 Jun '09

29 Jun '09

Hi I am using sipsak for following details... Below details are store in /tmp/mwi-immediate file NOTIFY sip:1212@mytest.com <sip%3A1212(a)mytest.com> SIP/2.0 From: <sip:asterisk@172.18.100.73 <sip%3Aasterisk(a)172.18.100.73>> To: <sip:1212@mytest.com <sip%3A1212(a)mytest.com>> Contact: <sip:1212@72.18.100.73 <sip%3A1212(a)72.18.100.73>> Call-ID: 023798(a)72.18.100.73 CSeq: 023798 NOTIFY Event: message-summary Content-Type: application/simple-message-summary Content-Length: 39 Messages-Waiting: yes Voicemail: 2/0 But when I execute sipsak command it gives following bad request #sipsak shoot -a number4 -f /tmp/mwi-immediate -s sip:1212@mytest.com<sip%3A1212(a)mytest.com>-v SIP/2.0 501 Method Not Supported Here Via: SIP/2.0/UDP 127.0.0.1:34389 ;branch=z9hG4bK.3242d1c3;rport=34389;alias;received=172.18.100.73 From: <sip:asterisk@172.18.100.73 <sip%3Aasterisk(a)172.18.100.73>> To: <sip:1212@mytest.com <sip%3A1212(a)mytest.com>> ;tag=d17c868d3732913d731aeb2b10d8dba7.1b83 Call-ID: 023798(a)72.18.100.73 CSeq: 023798 NOTIFY Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 Any idea... regarding the above problem.... -- Regards, Chandrakant Solanki

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[Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Uriel Rozenbaum 29 Jun '09

29 Jun '09

Guys, I've successfully upgraded schema and binaries for Kamailio, but when I start it, I get: Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:db_check_table_version: invalid version 2 for table carrierfailureroute found, expected 3 (check table structure and table "version") Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:carrierroute:carrierroute_db_init: during table version check. Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:init_mod: failed to initialize module carrierroute Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:main: error while initializing modules If I try and set 3 to the version table (for carrierfailureroute) I get the following: Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:db_check_table_version: invalid version 3 for table carrierfailureroute found, expected 2 (check table structure and table "version") Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:carrierroute:carrierroute_db_init: during table version check. Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:init_mod: failed to initialize module carrierroute Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:main: error while initializing modules Is there anything I have to do (appart from following the guidelines on http://kamailio.org/dokuwiki/doku.php/install:1.4.x-to-1.5.0)? Thanks! Uriel

3 8

[Serusers] auto-attendant problem
by IL'dar AKHmetgaleev 28 Jun '09

28 Jun '09

Greetings. I have a problem with auto-attendant. It accepts call immediately without playing my greeting message. How this problem can be fixed? My sip is akhilman(a)iptel.org -- Ильдар Ахметгалеев aka AkhIL Сбт Июн 27 23:52:48 KRAST 2009 Sat Jun 27 15:52:48 UTC 2009 ---------------------------------- Visit my home page http://akhilman.blogspot.com/ jabber: akhil(a)jabber.ru ---------------------------------- Маг формирует намерение совершить то, что он намерен совершить, просто за счет того, что он намеревается это совершить. (К.Кастанеда) ---------------------------------- Linux artstation 2.6.29-gentoo-r5 #1 SMP Mon Jun 1 19:50:26 KRAST 2009 x86_64 AMD Phenom(tm) 9550 Quad-Core Processor AuthenticAMD GNU/Linux up 12 days, 14:35, 1 user, load average: 0.20, 0.35, 0.38

2 2

[Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
by rubenrojas - Trc.es 26 Jun '09

26 Jun '09

Hello everyone, this is my first post on this list, I have installed kamailio 1.5.1 and set up a vanilla default kamailio.cfg, then I have modified the cfg to activate mysql, domain, presence, nathelper and authentication with md5, everything works as supposed to, and the clients can register, send txt messages and talk to each other. The only problem is with the audio when the two clients are behind a NAT, the phones can make a call and it does ring too, but when you pick up there is no audio both ways. when the phones have a public IP everything goes fine, it also works when I use a Linksys PAP2T whith the options to "Insert VIA received", "Insert VIA rport", "Handle VIA received", "Handle VIA rport" and "NAT mapping enable" turned on, with the Qutecom softphone works too. This is happening with thomson phones (model ST 2022), and GrandStream Budge Tone 200, it happens no matter what options I set for NATting on the phones, I've even used stun with stunserver.org or the ekiga stunserver, the phones register and can make and recieve calls, but there is no audio when you pick up the call. With a kamctl ul show, you can see that the phones have registered the Contact with their local IPs and the Received have the public IPs and ports for the NAT The only difference with the working Linksys is that they register the Contact with the public IP. Here you can see two NATed phones on the proxy Domain:: location table=512 records=2 max_slot=1 AOR:: 20000004(a)212.4.107.250 Contact:: sip:20000004@192.168.254.110:5060;transport=udp;user=phone Q= Expires:: 1150 Callid:: 72ed03f6d2f390f9(a)192.168.254.110 Cseq:: 10003 User-agent:: Grandstream BT200 1.1.6.27 Received:: sip:212.4.97.115:35379 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:212.4.107.250:5060 Methods:: 7807 AOR:: 20000000(a)212.4.107.250 Contact:: sip:20000000@192.168.254.101:5060;user=phone Q= Expires:: 2945 Callid:: 17fe-c0a80101-5-1(a)192.168.254.101 Cseq:: 6 User-agent:: THOMSON ST2022 hw2 fw3.56 00-18-F6-B5-7E-06 Received:: sip:212.4.97.115:55128 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:212.4.107.250:5060 Methods:: 4294967295 Im using rtpproxy and there is no log error that indicates that rttpproxy isn't working, in fact doing a SIP trace shows rtpproxy setting ports for the audio. I run rtpproxy with this command: rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F Any help would be greatly appreciated, I've been two weeks looking for a solution Im attaching my kamailio.cfg so you can take a look, at the end of the message Im gonna attache the SIP Trace of a call between two NATed phones (a Thomson and a GrandStream) in case anyone can help me decypher whats wrong here: this is my cfg file ************************************************************************************************** # # $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_mwi.so"#manually added loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 2) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #a#modparam("auth_db", "calculate_ha1", yes) #a#modparam("auth_db", "password_column", "password") #a#modparam("auth_db", "db_url", #a# "mysql://openser:openserrw@localhost/openser") #a#modparam("auth_db", "load_credentials", "") #parametros de autentificacion modificados manualmente modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "ha1") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 0) #modparam("auth_db", "use_domain", 0) modparam("auth_db", "use_domain", 1)#0 encendemos con 1 porque utilizaremos multi-domain modparam("auth_db", "load_credentials", "rpid") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:212.4.107.250:5060") # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@212.4.107.250") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) modparam("nathelper", "sipping_method", "OPTIONS") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } ##if (!uri==myself) /* replace with following line if multi-domain support is used */ if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) { route(2); } if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); setbflag(7);# sipping } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } # Auth only on registration #a# } else { #a# # authenticate if from local subscriber (uncomment to enable auth) #a# if (from_uri==myself) #a# { #a# if (!proxy_authorize("", "subscriber")) { #a# proxy_challenge("", "0"); #a# exit; #a# } #a# if (is_method("PUBLISH")) #a# { #a# if ($au!=$tU) { #a# sl_send_reply("403","Forbidden auth ID"); #a# exit; #a# } #a# } else { #a# if ($au!=$fU) { #a# sl_send_reply("403","Forbidden auth ID"); #a# exit; #a# } #a# } #a# #a# consume_credentials(); #a# # caller authenticated #a# } #a# } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ************************************************************************************************** ************************************************************************************************** And here goes the SIP Trace for a NATed to NATed hardphones: ************************************************************************************************** U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060 INVITE sip:20000000@212.4.107.250;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 332 v=0 o=20000004 8000 8000 IN IP4 192.168.254.110 s=SIP Call c=IN IP4 192.168.254.110 t=0 0 m=audio 40000 RTP/AVP 4 3 18 0 8 9 97 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:60 # U +0.000407 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668;rport=35379;received=212.4.97.115 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 # U +0.000034 212.4.107.250:5060 -> 212.4.97.115:55128 INVITE sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 348 v=0 o=20000004 8000 8000 IN IP4 192.168.254.110 s=SIP Call c=IN IP4 212.4.107.250 t=0 0 m=audio 35752 RTP/AVP 4 3 18 0 8 9 97 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:60 a=nortpproxy:yes # U +0.019311 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Content-Length: 0 # U +0.030480 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 # U +0.000083 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 # U +6.510103 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000365 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes # U +0.034122 212.4.97.115:35379 -> 212.4.107.250:5060 ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bKdf5e0ceed72f3797 Route: <sip:212.4.107.250;lr=on;nat=yes> From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone> Supported: path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 ACK User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.000245 212.4.107.250:5060 -> 192.168.254.101:5060 ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.2 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bKdf5e0ceed72f3797 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone> Supported: path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 ACK User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.458031 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000246 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes # U +0.999724 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000295 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes

5 5

[Kamailio-Users] NAThelper and multipart SDP
by Pascal Maugeri 26 Jun '09

26 Jun '09

Hi Does NATHelper support properly multipart SDP (eg. INVITE with application/sdp and application/resource-lists+xml) ? Regards, Pascal

12 53

[Kamailio-Users] Kamailio + Asterisk MWI Implementation
by Chandrakant Solanki 25 Jun '09

25 Jun '09

Hi All, Is there any link or document for implementing MWI (kamailio + asterisk) step by step.. I tried lot of searching but don;t found any such a good link or document. Please help me out. -- Regards, Chandrakant Solanki

4 5

[Serusers] Server dimensioning
by inge 25 Jun '09

25 Jun '09

Hi all, I wonder about the "child" parameter in ser.cfg. What's the average processing capacity (number of requets/process) ? Currently, the parameter is set to 4, the default value. What's the risk of the increase ? SER runs on RedHat AS4 with Intel Xeon Quad 3.0 Ghz. Thanks for your support.

2 2

[Kamailio-Users] new feature: PV class $T_inv(pv)
by Daniel-Constantin Mierla 25 Jun '09

25 Jun '09

Hello, by loading the tmx module, one can get access to INVITE attributes while processing the CANCEL, see more at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#t_inv_pv Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

2 1

[Kamailio-Users] LRN routing - 302 sip code
by Mik Cheez 24 Jun '09

24 Jun '09

One of our carriers gives us an option to check LRN routing via SIP before we attempt to send out a call. Our request will go out as a regular invite, then we'll get a 302 return code back, with the result contained in the header. My question is, how exactly is the flow of this kind of call supposed to work? I thought I could do a regular failover, but that doesn't work. Does anyone have a suggestion? Mik

3 4

[Kamailio-Users] script var or avp in ds_select_dst (dispatcher)?
by Robert McGilvray 24 Jun '09

24 Jun '09

Can someone please confirm that I cannot use a script var or avp in the set field when calling ds_select_dst from the dispatcher module? I'm trying to use dialplan to return a gateway group via attr, then call ds_select_dst with that but get the following error in my logs. If this is my problem are there any workarounds? The goal is to call ds_select_dst with a dynamic value. Example - ds_select_dst("$var(dpattr)", "0") Jun 24 14:50:49 [1125] ERROR:core:fixup_get_ivalue: no valid PV value found (error in scripts) Jun 24 14:50:49 [1125] ERROR:dispatcher:w_ds_select_dst: no dst set value If I hardcode the value which in this case is 30 it works as expected. I've verified that the correct value is in $var(dpattr) with an xlog right before the call to ds_select_dst. if ($ruri.user =~ "[0-9]{4}") { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu -> $ru - Internal 4 digit call\n"); if (dp_translate("2", "$ruri.user/$var(dpuser)")) { xlog("L_NOTICE", "KAM_HOST: ($ci) -> DP result='$var(dpuser)', attr='$var(dpattr)'\n"); setflag(FLAG_DIALPLAN); } else { xlog("L_NOTICE", "KAM_HOST: ($ci) -> No matching dialplan for $ru\n"); } } some other if statements here with other pattern matches... if (isflagset(FLAG_DIALPLAN)) { xlog("L_NOTICE", "KAM_HOST: ($ci) -> DP Attr is $var(dpattr)\n"); if (ds_select_dst("$var(dpattr)", "0")) { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu -> DST='$avp(s:dispatcher-dst)', \ GRP='$avp(s:dispatcher-grp)', CNT='$avp(s:dispatcher-cnt)'\n"); } else { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu -> $ru - No dst gateways found\n"); sl_send_reply("404", "No destination gateways found\n"); } } Thanks -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

1 1

[Kamailio-Users] SIP forking proxy
by Zeljko Lovrincevic 24 Jun '09

24 Jun '09

Hi, I need SIP forking proxy, and I would like to use OpenSER for that purpose. I have few questions if you could help. I am interesting for behavior when 200 OK INVITE is received for each forked dialog. Should only first 200 OK INVITE be passed to UA (the other will be terminated by proxy), or 200 OK for all forked dialogs should be passed to UA? Is it possibly to define some static configuration how to fork a dialogs or we always need REGISTER? Thanks in advance. BR/Zeljko

3 3

Re: [Kamailio-Users] [sr-dev] new version of lcr module
by Daniel-Constantin Mierla 24 Jun '09

24 Jun '09

Hi Juha, please add a short note in the wiki: https://sip-router.org/wiki/features/new-in-devel Makes easier to build new release notes and spread details about what is new in development version. Thanks, Daniel PS. I cc-ed users mailing list as this was a new feature announcement. On 06/22/2009 04:55 PM, Juha Heinanen wrote: > i just pushed a new version of lcr module, which resulted from > discussions on the mailing lists. > > most important new feature is support for any number of independent lcr > instances, which improves virtualization of sr. for example, different > domains can now manage their own least cost routing to their own gws. > > i also replaced aliveness check with new defunct_gw() function, which > script writer can use to defunct an unresponsive or busy gw for a > desired period of time. > > in order to keep lcr module manageable, i removed gw group related > functions (which were not efficient anyway). if someone is using them, i > hope that they can be replaced by lcr instances. > > -- juha > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev > > -- Daniel-Constantin Mierla http://www.asipto.com/

2 1

[Kamailio-Users] Failure at startup
by Francisco Javier Lizarán Vilches 24 Jun '09

24 Jun '09

Hi all: Running Openser v1.2.3-notls. It has been suffering failures at startup since some weeks ago and it seems to be related with database operation. Nothing has changed in config. The only remarkable change in the system is a slight increase in database registers at subscribers, location and lcr tables. Log excerpt from openser startup: Jun 19 15:23:32 openser[2196]: INFO: statistics manager successfully initialized Jun 19 15:23:32 openser[2196]: StateLess module - initializing Jun 19 15:23:32 openser[2196]: TM - initializing... Jun 19 15:23:32 openser[2196]: Maxfwd module- initializing Jun 19 15:23:32 openser[2196]: INFO:ul_init_locks: locks array size 512 Jun 19 15:23:32 openser[2196]: AUTH module - initializing Jun 19 15:23:32 openser[2196]: AUTH_DB module - initializing Jun 19 15:23:32 openser[2196]: TextOPS - initializing Jun 19 15:23:32 openser[2196]: AVPops - initializing Jun 19 15:23:32 openser[2196]: permissions - initializing Jun 19 15:23:32 openser[2196]: WARNING: File not found: /etc/openser/permissions.allow Jun 19 15:23:32 openser[2196]: Default allow file (/etc/openser/permissions.allow) not found => empty rule set Jun 19 15:23:32 openser[2196]: WARNING: File not found: /etc/openser/permissions.deny Jun 19 15:23:32 openser[2196]: Default deny file (/etc/openser/permissions.deny) not found => empty rule set Jun 19 15:23:32 openser[2196]: ACC - initializing Jun 19 15:23:32 openser[2196]: Dialog module - initializing Jun 19 15:23:32 openser[2196]: INFO: SNMPStats: Starting up the SNMPStats Module Jun 19 15:23:32 openser[2196]: UAC - initializing Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type (32) instead of <Refer-To> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type (7) instead of <Contact> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Referred-By> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Replaces> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Forwarder> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Referred-by> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <RG> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Origin> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Referred-By> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Replaces> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Forwarder> Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [DLG_status] Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [T_reply_code] Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type (4) instead of <From> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type (3) instead of <To> Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [T_reply_code] Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [T_reply_code] Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [T_branch_idx] Jun 19 15:23:32 openser[2196]: INFO: udp_init: SO_RCVBUF is initially 109568 Jun 19 15:23:32 openser[2196]: INFO: udp_init: SO_RCVBUF is finally 262142 ...Some registers, invites, etc logs... Jun 19 15:23:34 openser[2204]: new_connection: Can't connect to MySQL server on 'bbdd.bariksip.es' (4) Jun 19 15:23:34 openser[2201]: new_connection: Can't connect to MySQL server on 'bbdd.bariksip.es' (4) Jun 19 15:23:34 openser[2201]: ERROR:ul:child_init(5): Error while connecting database Jun 19 15:23:34 openser[2204]: ERROR:ul:child_init(-1): Error while connecting database Jun 19 15:23:34 openser[2201]: ERROR:init_mod_child: Error while initializing module usrloc, rank 5 Jun 19 15:23:34 openser[2204]: ERROR:init_mod_child: Error while initializing module usrloc, rank -1 Jun 19 15:23:34 openser[2201]: init_child failed Jun 19 15:23:34 openser[2204]: timer: init_child failed Jun 19 15:23:34 openser[2203]: INFO: signal 15 received Jun 19 15:23:34 openser[2201]: INFO: SNMPStats: The SNMPStats module got the kill signal Jun 19 15:23:34 openser[2201]: Shutting down the AgentX Sub-Agent! Jun 19 15:23:34 openser[2201]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill Jun 19 15:23:34 openser[2202]: INFO: signal 15 received Jun 19 15:23:34 openser[2200]: INFO: signal 15 received Jun 19 15:23:34 openser[2199]: INFO: signal 15 received Jun 19 15:23:34 openser[2198]: INFO: signal 15 received Jun 19 15:23:34 openser[2197]: INFO: signal 15 received Jun 19 15:23:34 openser[2204]: INFO: SNMPStats: The SNMPStats module got the kill signal Jun 19 15:23:34 openser[2204]: Shutting down the AgentX Sub-Agent! Jun 19 15:23:34 openser[2204]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill Jun 19 15:23:34 openser[2196]: new_connection: Can't connect to MySQL server on 'bbdd.bariksip.es' (4) Jun 19 15:23:34 openser[2196]: ERROR:ul:child_init(0): Error while connecting database Jun 19 15:23:34 openser[2196]: ERROR:init_mod_child: Error while initializing module usrloc, rank 0 Jun 19 15:23:34 openser[2196]: main: error in init_child Jun 19 15:23:34 openser[2196]: INFO: SNMPStats: The SNMPStats module got the kill signal Jun 19 15:23:34 openser[2196]: Shutting down the AgentX Sub-Agent! Jun 19 15:23:34 openser[2196]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill After multiple attempts the result is the same. This is the list of modules that use db: acc|auth_db|permissions|uri_db|usrloc|avpops|lcr|avpops|domain|speeddial|alias_db Looking at the traces, when openser places the request packet carrying this query "select prefix,from_uri,grp_id,priority from lcr", mysql server begins to perform retransmissions of the requested data; although openser eventually sends back the ack. Later request's responses from mysql server needs also retransmit until openser fails. If I change, for example, usrloc's dbmode to 0 (disables db use), the module that fails this time is auth_db. The mySql server is located in the same LAN that Openser(0.112s rtt). I have realized that Openser can be started without any problem when: -Log level is raised to 4(!). or -Another mysql server in a remote location is used (10s rtt). Has anybody experienced an issue like this? Thanks: Francisco Javier Lizarán Vilches ---------------------------------------------- BARIK (Grupo Ormazabal) Departamento Técnico Tel.: +34 917 479 900 www.barik.es <http://www.barik.es/> ----------------------------------------------

1 0

[Serusers] LCR module
by Ron 24 Jun '09

24 Jun '09

Hi All, Sorry about this post but i just can't figure out the logic: i'm using lcr to send calls to pstn, when my user dials a number that begins with 00 i re-route it to my termination providers. e.g.: if ((uri=~"^(sip:)?00[0-9]*@([a-z]+\.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@22\.22\.22\.22")) { strip(2); prefix("1234"); if(!load_gws()){ sl_send_reply("500", "Internal server error, unable to load gateways"); break; } } fortunately i have the same prefix on my termination providers. so on my lcr table i have defined all country codes to have the 1234 prefix, e.g 12341 for US , 123465 for singapore. etc. now i have a new provider, but prefix is different, how can i detect on lcr when all calls fail. i change prefix. i hope my question is clear sorry. regards, ron

1 0

[Serusers] LCR module
by Ron 24 Jun '09

24 Jun '09

Hi All, Sorry about this post but i just can't figure out the logic: i'm using lcr to send calls to pstn, when my user dials a number that begins with 00 i re-route it to my termination providers. e.g.: if ((uri=~"^(sip:)?00[0-9]*@([a-z]+\.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@22\.22\.22\.22")) { strip(2); prefix("1234"); if(!load_gws()){ sl_send_reply("500", "Internal server error, unable to load gateways"); break; } } fortunately i have the same prefix on my termination providers. so on my lcr table i have defined all country codes to have the 1234 prefix, e.g 12341 for US , 123465 for singapore. etc. now i have a new provider, but prefix is different, how can i detect on lcr when all calls fail. i change prefix. i hope my question is clear sorry. regards, ron

1 0

[Kamailio-Users] [REQ, $] db_rest (or db_http) module for database access
by Konstantin Cherkasov 23 Jun '09

23 Jun '09

Hello! Our company seeking for programmer who can create module for flexible database access. I call it db_rest (or db_http). This module must provide internal database API for other modules like db_auth. For access to external data it must use HTTPS-server and use REST-style data access (http://en.wikipedia.org/wiki/Representational_State_Transfer). Instead of making query like "SELECT f1,f2,f3 FROM tableA WHERE f4 = 'somevalue'" it must call HTTPS-server like this: GET http://hostname/database/tableA&c=f1,f2,f3&k=f4&op=eq&v=somevalue Server must return delimiter-separated data. Similarly (using POST) it must implement support for insert, update and so on. Of course we would like to see this module inside the main project repository. If you are interested please contact me for details (requirements, $$). -- Konstantin

4 17

[Kamailio-Users] Avoid encoding of CDR's written to database by Radius
by CSB 23 Jun '09

23 Jun '09

When a CDR is written that contains certain characters (e.g. +, ;) it is encoded (like a URL gets encoded). The SQL insert statement query is produced by OpenSER/Radius: 10625 Query INSERT INTO radacct (RadAcctId, AcctSessionId, AcctUniqueId, UserName, Realm, NASIPAddress, NASPortId, AcctStartTime, AcctStopTime, AcctSessionTime, AcctInputOctets, AcctOutputOctets, CalledStationId, CallingStationId, AcctTerminateCause, ServiceType, FramedProtocol, FramedIPAddress, AcctStartDelay, AcctStopDelay, SipResponseCode, SipMethod, SipTranslatedRequestURI, SipToTag, SipFromTag, SipRPID, SourceIP, SourcePort, CanonicalURI) VALUES (NULL, '149a4cbd403b071e109d89b50a17b3a5(a)202.180.1.1', 'e04d0b07ab2eca3e', SUBSTRING_INDEX('sip:094455667@202.180.1.1',':',-1), SUBSTRING_INDEX('sip:094455667@202.180.1.1', '@',-1), '127.0.0.1', '5060', '2009-06-20 11:44:49', '0', '0', '0', '0', SUBSTRING_INDEX('sip:=2B6012345678@147.202.1.1',':',-1), SUBSTRING_INDEX('sip:094455667@202.180.1.1',':',-1), '200', 'SIP', '', '', '0', '0', '200', 'Invite', SUBSTRING_INDEX('',':',-1), 'as03a3d864', 'as69938f56', '', '', '', SUBSTRING_INDEX('',':',-1)) However the records written to the file system by OpenSER/Radius are not encoded: Sat Jun 20 11:44:49 2009 Acct-Status-Type = Start Service-Type = SIP Sip-Response-Code = 200 Sip-Method = Invite Event-Timestamp = "Jun 20 2009 11:44:49 NZST" Sip-From-Tag = "as69938f56" Sip-To-Tag = "as03a3d864" Acct-Session-Id = "149a4cbd403b071e109d89b50a17b3a5(a)202.180.1.1" Called-Station-Id = "sip:+6012345678@147.202.1.1" Calling-Station-Id = "sip:094455667@202.180.1.1" User-Name = "sip:094455667@202.180.1.1" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "e04d0b07ab2eca3e" Timestamp = 1245455089 So how can I write +6012345678 to the database, rather than =2B6012345678? Any advice appreciated. Regards Cameron

2 2

[Serusers] SER Groups
by Shahbaz Ali 23 Jun '09

23 Jun '09

plz help me in setting the ser group mangment... how i add this feature....

1 0

[Kamailio-Users] Siputils error
by Carlos A. Alvarez 22 Jun '09

22 Jun '09

Hello, I keep getting this error when I try to setup ring_timeout ERROR:core:set_mod_param_regex: parameter <ring_timeout> not found in module <utils> I tried setting the modparam for siputils as well to no avail. I already read throught the documentation and I made sure that I am loading all pre requisites for this module. Also the command options is recorded wrong in the 1.5.1 documentation: modparam("options", "options_accept", "application/*") <-according to docs Looking through the source I found out that the correct command is modparam("siputils", "options_accept", "application/*") I wonder is the same is the issue with ring_timeout. Please help

5 8

[Kamailio-Users] Question about manually manipulating RURI
by Geoffrey Mina 22 Jun '09

22 Jun '09

I have a need to perform some manipulation of the user part of the RURI. I have some dialers which send INVITES with the RURI in the following format: d15555551212@123.345.123.345:5060 I need to check for the existence of the "d" and strip it off if it is present. Basically what I want is pseduo code is: if(left($ruri),1 == "d"){ $ruri = right($ruri,len($ruri)-1); } Thanks!

4 5

[Kamailio-Users] Cancel messes with ACC
by Mark Kent 22 Jun '09

22 Jun '09

Oops, found it... there is a typo in the example config file: # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; See the missing 's' on the table name... I guess I didn't see the error message when entering this. Thanks, -mark

2 1

[Kamailio-Users] Sending from 1 of N ip address
by Stagg Shelton 21 Jun '09

21 Jun '09

I have a requirement from an upstream provider that is providing multiple termination only gateways to me. The requirement is that I have to have a unique source ip address for each gateway that I will be terminating calls to. I would like to use only one kamailio system bound to multiple ip addresses to meet this requirement. I have my LCR built with the various routes to the gateways. I have identified the function force_send_socket([proto:]address[:port]) which seems like it can be used to meet this functional requirement. I am a little confused how to implement this function with LCR. My basic requirement is that if upstream gateway ip address == 1.2.3.4 then originate and maintain the dialog from my source ip 4.3.2.1 else if gateway ip == 6.7.8.9 then originate and maintain dialog using soure ip 9.8.7.6, else if ... N upstream gateways Is force_send_socket([proto:]address[:port]) the function that I need to be trying to use to meet this requirement? Is there a simpler method that can be used instead? Any help will be appreciated. Stagg

3 5

[Kamailio-Users] Cancel messes with ACC
by Mark Kent 20 Jun '09

20 Jun '09

I've been using openser 1.1.0 in a small pstn gateway config for a few years. I recently upgraded to kamailio 1.5.1. I started completely from scratch, with no DB, no users, and started with the example kamailio.cfg file. I made a few changes mostly at the bottom of the main request routing logic to ship things off to my pstn gateway, and re-entered things into the db. Works fine. Then I tried to enable accounting, with mysql, and this works when calls complete, but on a CANCEL I get this: kamailio[87208]: ACC: call missed: timestamp=1245456629;method=INVITE;from_tag=6343AD47-243F70A2;to_tag=2486;call_id=4529cab6-c7e08a51-dd349b2c(a)192.0.2.214;code=487;reason=Request Terminated;src_user=foouser;src_domain=sip.example.net;dst_ouser=18005551212;dst_user=18005551212;dst_domain=172.16.16.16 kamailio[87208]: ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'dst_user' in 'field list' kamailio[87208]: ERROR:core:db_do_insert: error while submitting query kamailio[87208]: ERROR:acc:acc_db_request: failed to insert into database My CANCEL processing is still the generic processing in the example kamailio.cfg file. Any hints? Thanks, -mark

1 0

Re: [Kamailio-Users] Question about manually manipulating RURI
by Carlos A. Alvarez 19 Jun '09

19 Jun '09

I don't know if this is what you are looking for, but I do something similar to process the From and To message based on a dialed prefix. Please note below my if statement and feel free to modify it for your own use: if ( search("To: <[Ss][Ii][Pp]:56011") ) { subst('/^To:(.*)sip:56011([0-9]+)@(.*)$/To:\1sip:+\2@\3/ig'); subst('/^From: "(.*)sip:([0-9]+)@(.*)$/From: "+\1sip:+\2@\3/ig'); }; On Fri, 2009-06-19 at 15:43 +0000, Geoffrey Mina wrote: Thanks, but I believe I just found the simpler way to strip, but I'm still not sure how I would check for the existence of the "d". http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#strip_int I need to do something like: if(left($ruri,1) == "d"){ //how do i write this statement? strip(1); } t_relay(); is that correct? thanks, Geoff On Fri, Jun 19, 2009 at 3:23 PM, Pascal Maugeri<pascal.maugeri(a)gmail.com<mailto:pascal.maugeri@gmail.com>> wrote: > Hi Geoffrey > > Why don't you try: > http://www.kamailio.org/docs/modules/1.5.x/textops.html#id2467793 > > Regards, > Pascal > > On Fri, Jun 19, 2009 at 9:01 PM, Geoffrey Mina <geoffreymina(a)gmail.com<mailto:geoffreymina@gmail.com>> > wrote: >> >> I have a need to perform some manipulation of the user part of the >> RURI. I have some dialers which send INVITES with the RURI in the >> following format: >> >> d15555551212(a)123.345.123.345<mailto:d15555551212@123.345.123.345>:5060 >> >> I need to check for the existence of the "d" and strip it off if it is >> present. Basically what I want is pseduo code is: >> >> if(left($ruri),1 == "d"){ >> $ruri = right($ruri,len($ruri)-1); >> } >> >> Thanks! >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org<mailto:Users@lists.kamailio.org> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org<mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] Dialog from 1 of N IP Addresses
by Stagg Shelton 19 Jun '09

19 Jun '09

I have a requirement from an upstream provider that is providing multiple termination only gateways to me. The requirement is that I have to have a unique source ip address for each gateway that I will be terminating calls to. I would like to use only one kamailio system bound to multiple ip addresses to meet this requirement. I have my LCR built with the various routes to the gateways. I have identified the function force_send_socket([proto:]address[:port]) which seems like it can be used to meet this functional requirement. I am a little confused how to implement this function with LCR. My basic requirement is that if upstream gateway ip address == 1.2.3.4 then originate and maintain the dialog from my source ip 4.3.2.1 else if gateway ip == 6.7.8.9 then originate and maintain dialog using soure ip 9.8.7.6, else if ... N upstream gateways Is force_send_socket([proto:]address[:port]) the function that I need to be trying to use to meet this requirement? Is there a simpler method that can be used instead? Any help will be appreciated. Stagg

2 1

[Kamailio-Users] [OT] SIP ALG Detector released
by Iñaki Baz Castillo 19 Jun '09

19 Jun '09

Hi, as part of my personal battle against SIP ALG routers, I've created an utility to detect such routers: http://dev.sipdoc.net/wiki/sip-stuff/SIP-ALG-Detector It has two parts: client and server: Basically, the client node running into the LAN sends an INVITE to the server node (running in a host with public IP). The request could be modified by the LAN router if SIP ALG is enabled. The server encodes the received request in Base64 and appends it to the SIP response. The client receives the response, decodes the body and gets the request as it arrived to the server. Then it creates a diff between the original request and the the request the server node received. These differences are displayed in the screen. Both, UDP and TCP, tests are performed. The client node is coded in Ruby and should work in any operating system (if Ruby is installed). The server node is also coded in Ruby. For those interested in try it, I have a server node running in my personal server: 87.98.230.161:5060 You can test the client against my server. Feedback is welcome :) Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 13

[Kamailio-Users] lcr - tag value NULL prefixed as "NULL"
by Andreas Granig 19 Jun '09

19 Jun '09

Hi, Kamailio 1.5: When having gw.tag column set to NULL in db (which seems to be allowed according to db schema), then next_gw() prefixes the R-URI user with the string "NULL". Setting it to empty string (''), nothing is prefixed. I'd expect that from a value of NULL as well. Could anyone please confirm this? Also, in http://kamailio.org/docs/modules/1.5.x/lcr.html the parameter prefix_mode is missing. It's available in README.lcr though, so probably the page is out of date? Andreas

2 3

[Kamailio-Users] AVP / Odd Error
by Brandon Armstead 19 Jun '09

19 Jun '09

Hey Guys, First off thanks for any help in advance! I'm adding a single avp test i.e. $avp(s:some-avp-name) = 'foo!'; to a development configuration running Kamailio revision: 5487 (1.4.3). When I start kamailio I'm getting the following error: CRITICAL:core:init_io_wait: could not alloc epoll array As soon as I delete this AVP addition and start her back up, she starts. Code Snippet (io_wait.c) 606 #ifdef HAVE_EPOLL 607 case POLL_EPOLL_LT: 608 case POLL_EPOLL_ET: 609 h->ep_array=local_malloc(sizeof(*(h->ep_array))*h->max_fd_no); 610 if (h->ep_array==0){ 611 LM_CRIT("could not alloc epoll array\n"); 612 goto error; 613 } 614 memset((void*)h->ep_array, 0, sizeof(*(h->ep_array))*h->max_fd_no); 615 if (init_epoll(h)<0){ 616 LM_CRIT("epoll init failed\n"); 617 goto error; 618 } 619 break; 620 #endif I do not know enough of Kamailio CORE but it looks like something to do with memory allocation / file descriptors? I've increased available memory to kamailio as well as open file limits / file descriptors etc -- to no avail, any help is much appreciated thanks guys!

2 5

[Serusers] (no subject)
by Mohammed Fareed 18 Jun '09

18 Jun '09

Hi, kindly add as server user thanks Regards, V.P.Mohammed Fareed Analyst Programmer Mobile : +966-5-592303381 KSA. Cricket on your mind? Visit the ultimate cricket website. Enter http://cricket.yahoo.com

1 0

[Kamailio-Users] Multipart SDP problem...
by Edson - Lists 18 Jun '09

18 Jun '09

Hi, Guys... After trying a PSTN call through an AS5300-Cisco GW got this error: ==== ON-REPLY[1] incoming reply <200> for <INVITE> <8999>/<XXXXXXXXXXX> ERROR:nathelper:check_content_type: invalid type for a message ERROR:nathelper:extract_body: content type mismatching ERROR:nathelper:force_rtp_proxy: can't extract body from the message == Googling around I found an e-mail from Ovidus (23/12/2008) telling that a multi part SDP wasn't integrated on Nathelper. I suppose that I faced such an issue. How to circumvent/workaround this error message? As a side effect call gives no RTP flow, since one direction send it to RTPProxy and the other directly. Edson. -x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x Relevant info.... Using: ==== # kamailio -V version: kamailio 1.5.1-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, USE_SCTP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:5868M @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 13:54:27 Jun 4 2009 with gcc 4.3 == The ON-REPLY code is: ==== onreply_route[1] { xlog("L_INFO","ON-REPLY[1] incoming reply <$rs> for <$rm> <$fU>/<$tU>"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } == The ofending SDP in case is: ==== Content-Type: multipart/mixed;boundary=uniqueBoundary. Content-Length: 378. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7829 2204 IN IP4 XXX.YYY.ZZZ.10. s=SIP Call. c=IN IP4 XXX.YYY.ZZZ.10. t=0 0. m=audio 16976 RTP/AVP 0. c=IN IP4 XXX.YYY.ZZZ.10. a=rtpmap:0 PCMU/8000. --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . ANM,. PRN,isdn*,,QSIG*,. . --uniqueBoundary--. ==

8 32

Re: [Kamailio-Users] Trying to make Purple Module work
by Eric PTAK 18 Jun '09

18 Jun '09

Hi mani, Looking your log, I see you're trying to contact sip:manivasagam.s*msn.com@ kmilo.smithmicro.com as I mentionned to be a possible way to get purple working without mapping non-sip friends. But it requires modifications in the purple_* functions to support it. Then, there is also the usrloc error cause it didn't find such user. You must use a purple_gateway_domain different from your main domain, in order to filter messages to the gateway easyly, priori to other routes. for example (but again the module need mods) : route { # initial sanity check, loopavoidance & big messages ..... ..... # retransmission absorbtion ..... ..... # messages to purple gateway if ($rd == "purple_gateway_domain") { route(10); } # main route ..... ..... } #purple route route[10] { #method filtering to call the appropriate purple function } Eric. 2009/6/16 mani sivaraman <mani.opensips(a)gmail.com> > Attached is kamailio debug log while trying to send a MESSAGE to an MSN > contact from my SIP Account. manivasaga.s(a)msn.com is the MSN account > available on the SIP Buddy list. THe MESSAGE send fails and the server sends > back 202 and then 404. > > THe debug message is attached. Any input is appreciated. > > On Tue, Jun 16, 2009 at 10:00 AM, mani sivaraman <mani.opensips(a)gmail.com>wrote: > >> Hi Eric >> Could you please post a working example kamailio.cfg file with the example >> purplemap table contents (remove passwords from it). I guess I'm doing some >> mistake in coding purple_handle_subscribe, purple_handle_publish() and >> purple_send_message. Your help is really apppreciated. >> >> What IM Client did you use to test/develop purple ? Is it X-Lite ? I would >> like to try purple with that client first. Please let me know. I want to get >> the first PUB/SUB and MESSAGE working bet SIP and MSN or Gmail or yahoo. >> >> Your help is appreciated. >> - Mani >> >> >> On Tue, Jun 16, 2009 at 2:56 AM, Eric PTAK <eric.ptak.fr(a)gmail.com>wrote: >> >>> Hi, >>> >>> The PUBLISH has to be processed normally by the Presence >>> Server (eg handle_publish) to get the response. >>> Then, you can also use purple_handle_publish to propagate you status to >>> external protocols. >>> Whenever, you should not get that error regarding the db entry and the >>> publish sip uri... >>> Need more checks... >>> >>> Eric. >>> >>> 2009/6/15 mani sivaraman <mani.opensips(a)gmail.com> >>> >>>> Hi Eric >>>> I keep getting this error messages when I setup the purplemap table in >>>> mysql and enable purple PUB/SUN processing in openser. I have only one >>>> mapping purplemap table like this. My SIP client is trying to send presence >>>> PUBLISH, but this never gets ACKed by openser server. And I get this error >>>> every time PUB happens "ERROR:purple:find_accounts: result count = 0" >>>> >>>> sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>| >>>> smvasagam2000(a)gmail.com | gtalk | xxxx >>>> >>>> >>>> PUBLISH sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>from [ >>>> sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>] >>>> to [sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com> >>>> ] >>>> Jun 15 14:37:47 [17541] ERROR:purple:find_accounts: result count = 0 >>>> SUBSCRIBE sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>from [ >>>> sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>] >>>> to [sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com> >>>> ] >>>> Jun 15 14:37:47 [17535] INFO:presence:send_notify_request: NOTIFY >>>> sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>via sip:msivaraman@172.16.1.125:6332;transport=udp >>>> on behalf of sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>for event presence.winfo >>>> SUBSCRIBE sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>from [ >>>> sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com>] >>>> to [sip:msivaraman@kmilo.smithmicro.com<sip%3Amsivaraman(a)kmilo.smithmicro.com> >>>> ] >>>> Jun 15 14:37:47 [17541] ERROR:purple:find_accounts: result count = 0 >>>> >>>> Your assistance is appreciated. >>>> Thanks >>>> Mani >>>> >>>> >>>> On Mon, Jun 15, 2009 at 11:13 AM, mani sivaraman < >>>> mani.opensips(a)gmail.com> wrote: >>>> >>>>> That was it. I did that first time and then I forgot to set the env >>>>> variable again. I will add it to ctl script now. Thanks for your help. I >>>>> appreciate it. I will now go on to adding processing of MESSAGE, PUBLISH >>>>> NOTIFY etc. >>>>> >>>>> Thank you >>>>> >>>>> >>>>> On Mon, Jun 15, 2009 at 10:14 AM, Eric PTAK <eric.ptak.fr(a)gmail.com>wrote: >>>>> >>>>>> Did you added LD_PRELOAD=/path/to/your/libpurple.so env var in your >>>>>> ctl script ? >>>>>> >>>>>> Eric. >>>>>> >>>>>> 2009/6/15 mani sivaraman <mani.opensips(a)gmail.com> >>>>>> >>>>>> Thanks for getting back Eric. After opensips I tested the latest >>>>>>> Kamailio 1.5.1 code with purple module. I get the same SIGCHLD core dump on >>>>>>> Kamailio as well. I saw Kamailio comes with purple source code. If I load >>>>>>> purple.so and then add the modprobe for purple, I get the core dump. Is >>>>>>> there any thing I can check or look for in the debug to make sure it works. >>>>>>> If I unload purple and remove the respective modprobes every thing seems to >>>>>>> work fine. >>>>>>> >>>>>>> Thank you >>>>>>> >>>>>>> >>>>>>> On Mon, Jun 15, 2009 at 3:15 AM, Eric PTAK <eric.ptak.fr(a)gmail.com>wrote: >>>>>>> >>>>>>>> Hi mani, >>>>>>>> >>>>>>>> purple module hasn't been tested with opensips, and I can't say if >>>>>>>> it should work. >>>>>>>> but just a question : how did you compile the module ? eg, within >>>>>>>> which project ? >>>>>>>> >>>>>>>> regards, >>>>>>>> eric. >>>>>>>> >>>>>>>> 2009/6/12 mani sivaraman <mani.opensips(a)gmail.com> >>>>>>>> >>>>>>>>> I compiled and installed opensips 1.5.1 with mysql db and works >>>>>>>>> fine. I compiled and installed the new purple module from outside and >>>>>>>>> installed the same. if I try to load the purple.so module and add modprobe >>>>>>>>> for purple params, opensips core dumps saying >>>>>>>>> >>>>>>>>> Jun 12 12:06:38 [22121] CRITICAL:core:receive_fd: EOF on 15 >>>>>>>>> Jun 12 12:06:38 [22103] INFO:core:handle_sigs: child process 22110 >>>>>>>>> exited by a signal 11 >>>>>>>>> Jun 12 12:06:38 [22103] INFO:core:handle_sigs: core was generated >>>>>>>>> Jun 12 12:06:38 [22103] INFO:core:handle_sigs: terminating due to >>>>>>>>> SIGCHLD >>>>>>>>> Jun 12 12:06:38 [22104] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22105] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22106] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22107] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22108] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22114] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22109] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22115] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22116] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22117] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22119] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22121] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22112] INFO:core:sig_usr: signal 15 received >>>>>>>>> Jun 12 12:06:38 [22103] NOTICE:presence:destroy: destroy module ... >>>>>>>>> >>>>>>>>> >>>>>>>>> The gdb back trace is not very useful either. >>>>>>>>> >>>>>>>>> (no debugging symbols found) >>>>>>>>> Core was generated by `opensips'. >>>>>>>>> Program terminated with signal 11, Segmentation fault. >>>>>>>>> [New process 22083] >>>>>>>>> #0 0x085a9c6d in ?? () >>>>>>>>> (gdb) backtrace full >>>>>>>>> #0 0x085a9c6d in ?? () >>>>>>>>> No symbol table info available. >>>>>>>>> (gdb) >>>>>>>>> >>>>>>>>> Help Any one , Please. >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Users mailing list >>>>>>>>> Users(a)lists.opensips.org >>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>> >>>> >>> >> >

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Re: [Kamailio-Users] disable_dns_blacklist
by Karl Smith 18 Jun '09

18 Jun '09

After the second attempt that Kamailio sends a call towards an ip address and gets a negative sip response say 503, kamailio will block that ip route for approx 4 minutes and sends 401 unauthorized immediately to the sip client. That ip route can be a gateway which is sending a negative reponse for certain destinations only and this may be due to a problem which is beyond the gateway itself; so I want to avoid blocking the complete route automatically. It appears that, in version 1.5 the route gets blocked for a longer period. Appreciate any help. Thanks, Karl --- On Thu, 18/6/09, Iñaki Baz Castillo <ibc(a)aliax.net> wrote: From: Iñaki Baz Castillo <ibc(a)aliax.net> Subject: Re: [Kamailio-Users] disable_dns_blacklist To: Cc: users(a)lists.kamailio.org Date: Thursday, 18 June, 2009, 3:07 PM 2009/6/18 Karl Smith <sipware(a)yahoo.co.uk>: > Hi, > > I have always used the global setting disable_dns_blacklist=yes so as to > avoid openser from temporarily blocking a particular route that may return a > negative sip response such as 503. > > I am now planning a migration to kamailio but the feature does not seem to > be working. I have tried both kamailio 1.4 and also 1.5. Since Kamailio 1.4 (perhaps since OpenSer 1.3) this option is dissabled by default (disable_dns_blacklist=yes). What do you mean with "it doesn't work"? -- Iñaki Baz Castillo <ibc(a)aliax.net> _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] disable_dns_blacklist
by Karl Smith 18 Jun '09

18 Jun '09

Hi, I have always used the global setting disable_dns_blacklist=yes so as to avoid openser from temporarily blocking a particular route that may return a negative sip response such as 503. I am now planning a migration to kamailio but the feature does not seem to be working. I have tried both kamailio 1.4 and also 1.5. Any ideas? Thanks, Karl

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[Kamailio-Users] disable_dns_blacklist
by Karl Smith 18 Jun '09

18 Jun '09

Hi, I have always used the global setting disable_dns_blacklist=yes so as to avoid openser from temporarily blocking a particular route that may return a negative sip response such as 503. I am now planning a migration to kamailio but the feature does not seem to be working. I have tried both kamailio 1.4 and also 1.5. Any ideas? Thanks, Karl

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[Kamailio-Users] Q value with alias_db?
by Robert McGilvray 18 Jun '09

18 Jun '09

This seems like it should be easy but based on the database schema I can't figure out how I can specify a Q value for different destinations when using alias_db with mysql. I need to map external DIDs to phones registered to kamailio and some to other gateways, sometimes those numbers will need to fork based on the q. I can probably do what I need with direct sql queries I just want to make sure I'm not missing something with db_aliases first. Thanks -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Number portability
by Paul Mancheno H. 18 Jun '09

18 Jun '09

Hello friends I want that a call out of my softswitch go to my Kamailio, there run a query to a database, and return to the softswitch preceded with an index which I will forward the proper operator, the most important thing is that only pass signaling to Kamailio, and only when the call is going to do is send the data packets, and ultimately the data come from softswitch to softswitch. Should I make any special configuration for this to happen? A lot of thanks.

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[Kamailio-Users] ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN
by Andre Warnier 18 Jun '09

18 Jun '09

Hi all .. When I send a register packet to my kamctl, then I using " kamctl ul show "openser" " to see success or not with SIP server , it's response ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN Do I have defind SIP_DOMAIN in /usr/local/etc/kamailio/ kamctlrc ? SIP_DOMAIN is mean IP that install kamctl ? Any help would be greatly appreciated Thanks & regards, Andre

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[Serusers] SER.cfg problem
by Shahbaz Ali 17 Jun '09

17 Jun '09

plz help I attach my ser.cfg when i enter ser -c the error is: " 0(5976) parse error (150,18-19): unknown command, missing loadmodule? ERROR: bad config file (1 errors)" Best regards Shazi

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[Kamailio-Users] Pass a variable to config script
by sasirekha jaganathan 17 Jun '09

17 Jun '09

Hi, I need to pass a variable from kamailio code base to the configuration script. Can I add a new parameter to the existing pre script callbacks? Is there any other method to achieve the same? Thanks Love Cricket? Check out live scores, photos, video highlights and more. Click here http://cricket.yahoo.com

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Re: [Kamailio-Users] FW: ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN
by Saúl Ibarra 17 Jun '09

17 Jun '09

No, it's created when you start Kamailio... have you restarted after making the changes? PS: Again, please, reply to the mailing list... -- Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes." ---------------------------------------------------------------- http://www.saghul.net/

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[Kamailio-Users] Kamailio Core Dumping
by Stagg Shelton 17 Jun '09

17 Jun '09

I've recently tried to migrate my production environment to kamailio 1.5.1-notls from openser-1.3.4-notls. I have been working through some issues, and the most recent issues have been that kamailio has begun to segmentation fault and core dump. Has anyone else experienced any similar issues, what other information could i provide that can be helpful in diagnosing my troubles with 1.5.1. I have had to fall back to 1.3.4 until i can get this sorted out. Below is my backtrace of the core dump file. Core was generated by `/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. [New process 5315] #0 0x00007f57bb9954e0 in update_dialog_dbinfo (cell=0x7f577e6a17c0) at dlg_db_handler.c:498 498 SET_STR_VALUE(values+8, cell->bind_addr[DLG_CALLEE_LEG]- >sock_str); Missing separate debuginfos, use: debuginfo-install bzip2- libs-1.0.5-3.fc10.x86_64 compat-db45-4.5.20-5.fc10.x86_64 e2fsprogs- libs-1.41.4-5.fc10.x86_64 elfutils-libelf-0.141-1.fc10.x86_64 file- libs-4.26-5.fc10.x86_64 glibc-2.9-3.x86_64 keyutils- libs-1.2-3.fc9.x86_64 krb5-libs-1.6.3-18.fc10.x86_64 libconfuse-2.6-1.fc9.x86_64 libgcc-4.3.2-7.x86_64 libselinux-2.0.78-1.fc10.x86_64 lm_sensors-3.0.2-1.fc10.x86_64 lua-5.1.4-1.fc10.x86_64 mysql-libs-5.0.77-1.fc10.x86_64 net-snmp- libs-5.4.2.1-3.fc10.x86_64 nspr-devel-4.7.3-2.fc10.x86_64 nss- devel-3.12.2.0-5.fc10.x86_64 openssl-0.9.8g-13.fc10.x86_64 pcre-7.8-1.fc10.x86_64 perl-libs-5.10.0-68.fc10.x86_64 popt-1.13-4.fc10.x86_64 radiusclient-ng-0.5.6-3.fc9.x86_64 rpm- libs-4.6.0-2.fc10.x86_64 tcp_wrappers-libs-7.6-53.fc10.x86_64 zlib-1.2.3-18.fc9.x86_64 (gdb) bt #0 0x00007f57bb9954e0 in update_dialog_dbinfo (cell=0x7f577e6a17c0) at dlg_db_handler.c:498 #1 0x00007f57bb998b5e in dlg_onreply (t=0x7f577aaec3b8, type=<value optimized out>, param=<value optimized out>) at dlg_handlers.c:318 #2 0x00007f57bf59d345 in run_trans_callbacks (type=128, trans=0x7f577e6a3918, req=0x7f577e6afce8, rpl=0x775710, code=200) at t_hooks.c:240 #3 0x00007f57bf5ba352 in relay_reply (t=0x7f577e6a3918, p_msg=<value optimized out>, branch=0, msg_status=200, cancel_bitmap=0x7fffc7c0c098) at t_reply.c:1229 #4 0x00007f57bf5badce in reply_received (p_msg=0x775710) at t_reply.c: 1485 #5 0x0000000000420382 in forward_reply (msg=0x775710) at forward.c:521 #6 0x00000000004529f9 in receive_msg ( buf=0x71f880 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 8.17.32.37;branch=z9hG4bKd89d.68915ca2.0\r\nVia: SIP/2.0/UDP 8.17.32.166 :5060;received=8.17.32.166;branch=z9hG4bK0615b7d2;rport=5060\r\nFrom: \"CID Info\" <sip:5551212"..., len=889, rcv_info=0x7fffc7c0c190) at receive.c:212 #7 0x0000000000487da6 in udp_rcv_loop () at udp_server.c:449 #8 0x00000000004274e4 in main_loop () at main.c:774 #9 main (argc=<value optimized out>, argv=0x7fffc7c0c368) at main.c: 1321 (gdb)

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[Kamailio-Users] Access to $var() from perl
by Alex Hermann 17 Jun '09

17 Jun '09

Hello, (how) can I access $var() variables via the perl module? OpenSER::AVP::get() does not work for $var(). -- Greetings, Alex Hermann

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Re: [Kamailio-Users] FW: ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN
by Saúl Ibarra 17 Jun '09

17 Jun '09

On Wed, Jun 17, 2009 at 8:45 AM, André Warnier<andrewarnier(a)gmail.com> wrote: > Hi Saúl : > I have set SIP_DOMAIN=142.11.20.10 in /usr/local/etc/kamailio/kamctlrc , > Then when I using kamctl ul show openser(a)142.11.20.10 , I got the error as > follows : > > ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > "/tmp/kamailio_fifo")' in your config > ERROR: and also have loaded the mi_fifo module. > > What's Wrong With This? > Just read the error, it's very descriptive :) You need to setup the fifo so that kamctl can communicate with the kamailio process. PS: Please, reply to the mailing list, no to me. PPS: Don't crosspost!! -- Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes." ---------------------------------------------------------------- http://www.saghul.net/

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[Kamailio-Users] Error opening Kamailio's FIFO /tmp/kamailio_fifo
by vinod 17 Jun '09

17 Jun '09

Hello, When I using " kamctl ul show "openser" " , I got the error : ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. What's Happening!! Thanks in advance, Vinod

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[Kamailio-Users] FW: ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN
by Andre Warnier 17 Jun '09

17 Jun '09

Hi all .. When I send a register packet to my kamctl, then I using " kamctl ul show "openser" " to see success or not with SIP server , it's response ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN Do I have defind SIP_DOMAIN in /usr/local/etc/kamailio/ kamctlrc ? SIP_DOMAIN is mean IP that install kamctl ? Any help would be greatly appreciated Thanks & regards, Andre

3 2

[Kamailio-Users] ACK/BYE problem with NAT
by Dubravko Caric 16 Jun '09

16 Jun '09

Hi all, I have a problem when calling from UAC#1 (public IP address) to UAC#2 (private IP address), call is dropping after 30 sec. STUN and ICE are disabled in UAC#2. Other way is working OK (UAC#2 --INVITE--> UAC#1), RTPProxy is in the middle,... Kamailio is aware that UAC#2 is behind NAT and adds received filed on REGISTER. MySQL location table: contact received sip:bob@10.2.4.18:1025;rinstance=24c6e191484e570a sip:UAC#2_PUBLIC_IP_ADDRESS:55671 Why proxy is not aware that UAC#2 is behind NAT (for BYE and ACK) when call is initiated by UAC#1 and thus sends ACK/BYE to private IP address? Shouldn't Contact header be rewriten with NAT address/port and also a "c" filed in SDP with RTPProxy address? fix_nated_contact() sholud fix contact header for transaction-starting messages like ACK and BYE... I'm using kamailio 1.5.1-tls and standard kamailio.cfg (to be sure that I'm not using my cfg which could be faulty)... Thanks a lot Dubravko INVITE: *working OK* # U UAC#1_PUBLIC_IP_ADDRESS:24354 -> KAMAILIO_PUBLIC_IP_ADDRESS:5958 INVITE sip:bob@example.com SIP/2.0.... U UAC#1_PUBLIC_IP_ADDRESS:5958 -> UAC#2_PUBLIC_IP_ADDRESS:55671 INVITE sip:bob@10.2.4.18:1025;rinstance=24c6e191484e570a SIP/2.0.. ACK (and BYE): *not working* # U UAC#1_PUBLIC_IP_ADDRESS:24354 -> KAMAILIO_PUBLIC_IP_ADDRESS:5958 ACK sip:bob@10.2.4.18:1025;rinstance=24c6e191484e570a SIP/2.0.... # U KAMAILIO_PUBLIC_IP_ADDRESS:5958 -> UAC#2_PRIVATE_IP_ADDRESS:1025 ACK sip:bib@10.2.4.18:1025;rinstance=24c6e191484e570a SIP/2.0....

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