- sr-users - kamailio.org

Sip express live Cd router
by Shoukat, Kashif 28 Jan '13

28 Jan '13

Dear , Can you please provide me link of sip express live Cd router .. I want to use it... waiting your kind response Regards, Kashif Shoukat | Analyst Software Engineer | Work : + 1.202.289.9898 x 9077 Mobile : + 92.323.431.8548 Skype: kashif.shoukat.dgs www.dgsworld.com<http://www.dgsworld.com/> dgs deliberate by design This message contains information which may be confidential and privileged. Unless you are the addressee (or authorized to receive for the addressee), you may not use, copy or disclose to anyone the message or any information contained in the message. If you have received the message in error, please advise the sender by replying to kashif.shoukat(a)dgsworld.com<mailto:kashif.shoukat@dgsworld.com> and delete the message.

1 0

TLS certificate chain Android verification
by Nord7 28 Jan '13

28 Jan '13

Hello all. Our company buy Thawte SSL Web server certificate. I'm enabled in Kamailio 3.3 TLS support and configured him. Adding to "myhost.pem" certificate, two "intermediate" certificates (like in chain) "cat myhost.pem SSL_SecondaryCA.pem SSL_PrimaryCA.pem > chain-cert.pem". Remark: I use this chain on apache2. And Android trusted to this certificate chain. After this i use chain-cert.pem with KAMAILIO 3.3 and SIP client "BRIA Android". "Bria Android" says certificate is not trusted. Next step "BRIA IPHONE" and certificate chain works (trusted)! BRIA support says me - BRIA don't trusts certificate if system don't trusts. Why Android don't trusts to my chain-certificates with KAMAILIO? If his trust to same certificate with Apache? I'm understand this is not Kamailio problem, but maybe any one who have expirience in this problem help me? I know about bouncycastle, but this is not best decision for me. Maybe something else can help? I tryed this with Android 2.3.7, 4.0, 4.1. Best regards Kirill.

1 0

event_route for pre-dialog start
by Mino Haluz 28 Jan '13

28 Jan '13

Hi, I think that it is more than likely not implemented, but is there any event-route that is triggered just before event_route[dialog:start] ? I need to check some security flags before the dialog is created. But it is too early too check them in relay route.. Mino

1 0

Is kamailio creating multiple connections to db?
by Mino Haluz 28 Jan '13

28 Jan '13

Hi, I made some DB test on our HA cluster, and I got that for 1 process I have 1000 queries/sec and for 11 processes 3000 queries/sec. That's why I would to ask if every kamailio process that is spawned has its own database connection or queries are put in the queue and executed by some main process ? I would like to know this, so that I could assess maximum call count/sec that db can handle. Thank you, Mino

3 4

Generating logging based on SIP messages
by Grant Bagdasarian 28 Jan '13

28 Jan '13

Hello, A while ago I came across the siptrace and sipcapture modules and thought this would be a good way to generate logging out of all the SIP messages that are passed through Kamailio and stored in the siptrace/sipcapture table. My idea was to extract certain values from SIP messages and store these in our database. Also each SIP Request or Response would be mapped to a certain status so we would be able to monitor which call is in which state. I still have a lot of work to do to think this all out, since it's still only an idea. I'm just wondering if anyone could give me some pointers into what to keep in mind when doing this. Regards, Grant

2 2

Pseudo variable to get instance in the contact....
by Krishna Kurapati 28 Jan '13

28 Jan '13

Hi, I added in my local branch "$ct.inst" pseudo variable to retrieve instance. Since instance could be an important attribute to make routing decisions, having quick access to it in the message may be useful. Anyone else need this functionality? Krish Kura

3 2

iptel x media5-fone problem
by Kasper Hämäläinen 28 Jan '13

28 Jan '13

hi guys, hope u got time to help me. Im first timer testing VoIP and SIP. I created account to ur iptel.org site. I downloaded "Jitsi"-software. First when I opened it, I login with my username and password. Then I go to "add new account", give same login username and password, but Jitsi says that my password is wrong..wtf. And I installed Media5-Fone to my iPhone 4S. When adding SIP-account it's says again that I got wrong password. And YES, I didn't mistype it. Thanks - Kasper

2 1

CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg - Kamailio 3.3.x
by Rinor Hoxha 28 Jan '13

28 Jan '13

Hi list, I'm having some issues with dialog module. From time to time I get the following error: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) [root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" | sort | uniq -c 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil) This one was flooding my log file: tl=0x2aeeea1c6840 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this happening and how to fix it. (I suspect that may be some issues with our server memory...checking now...and will inform). However wanted to raise this issue in case anyone else is having the same. Thanks, Rinor

2 1

Pseudo variable to access SIP instance in contact
by Krishna Kurapati 28 Jan '13

28 Jan '13

Hi, Is there a way to directly access the SIP instance present in message from the script? Thanks Krish Kura

2 1

Kamailio behind NAT - best practice
by Klaus Darilion 27 Jan '13

27 Jan '13

Hi! Up to now I could avoid Kamailio setups with Kamailio behind NAt. But this time I have to deal with it. That's why I want to ask what others did as best practice. The scenario is: Asterisk 1----\ Kamailio+RTPPROXY \ |10.10.0.2 Asterisk n------\ | --------|------------- FW --SIP-trunk--- ITSP Freeswitch 1----/ 10.10.0.1 public-IP / Freeswitch n--/ 10.10.0.x Kamailio and rtpproxy have a private IP. Internal communication uses private IPs, external communication uses a public IP which is NATed 1:1 to Kamailio's IP address. No registrations, just forwarding of messages. Using the global "advertised_address" setting with the public IP does not work, as there is also internal communication. Using set_advertised_address() is also cumbersome. So it seems, the easiest solution would be to use 2 sockets on Kamailio, e.g. port 5050 and port 5060. Then I could use the listen with dedicated advertised addresses: listen=udp:10.10.0.2:5050 listen=udp:10.10.0.2:5060 advertise pu.bl.ic.ip:5060 If I understand it correctly, this should solve all issues with Record-Routing and Via-headers. For RTP-Proxy it seems necessary to detect the direction of each message and set the IP address in rtpproxy_manage("","ip.add.re.ss") manually. Thus, it seems straight forward - or do I miss something? Any comments and practical experience? Thanks Klaus

5 8

Pseudo variables and uid_domain module compatibility
by Philippe Sultan 25 Jan '13

25 Jan '13

Hi, According to docs of "uid_domain.so", $fd and $td are attribute groups which are set when calling the lookup_domain function, to retrieve domain attributes stored in an DB ($fd.did or $td.did) : lookup_domain("$fd", "@from.uri.host"); lookup_domain("$td", "@ruri.host"); if (strempty($fd.did) && strempty($td.did)) { # Neither the calling nor the called domain is local # This is a relaying attempt which should be forbidden sl_reply("403", "Relaying Forbidden"); drop; } But then, starting Kamailio fails with a configuration file error : 0(2098) ERROR: <core> [pvapi.c:761]: error searching pvar "fd" 0(2098) ERROR: <core> [pvapi.c:965]: wrong char [d/100] in [$fd] at [2 (0)] 0(2098) : <core> [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 332, column 15-17: unknown script pseudo variable $fd ERROR: bad config file (1 errors) Line 332 is : if (strempty($fd.did) && strempty($td.did)) { I thought I could start Kamailio without "pv.so" (and use selects instead), but the error still shows up, as it is reported by the core pvapi.c file. So I'm kind of stuck, and asking help from the community to fix this problem :D Philippe

2 8

uac_replace_from recovery with modified header
by Andreas Granig 25 Jan '13

25 Jan '13

Hi, This is a question for kamailio 3.0.x and I'm still trying to reproduce this issue somehow for latest stable, but maybe it was a known issue anyways which got fixed in newer versions, so I don't have to dig deeper into this: There is an INVITE "A -> kamailio -> B", and kamailio does uac_replace_from() to replace the display- and user-part (leaving the domain-part untouched). When there's a BYE "B -> kamailio -> A", then From/To are switched, so the replaced From of the INVITE becomes the replaced To of the BYE. Kamailio gets that one right and tries to recover (in auto-mode) the To. However, in this BYE, B changes the domain-part of the To, which should be valid, as elements are only supposed to inspect the tags of From/To, right? The problem though is that when recovering the To, kamailio inserts garbage into the domain part. The vsf-Parameter in the Route of the BYE is definitely ok though, so that doesn't seem to be the issue. Here's an example: In the INVITE, From: "ab12345"<sip:ab12345@192.168.0.12;user=phone>;tag=948a3340 gets rewritten to From: "12345"<sip:12345@192.168.0.12;user=phone>;tag=948a3340 And in the BYE, To: "12345"<sip:12345@1.2.3.4>;tag=948a3340 gets rewritten to To: "12345"<sip:ab12345<@-%:68*0.0%12;user=phone>;tag=948a3340 Note that the received To in the BYE has a different host-part than what got sent out in the From of the INVITE, and when kamailio recovers the To, it doesn't recover the Display-part (not sure though if this is intended behaviour anyway), but it only party recovers the domain-part. You can still see some artefacts of the original domain (like the "68" from the second "168" octet, and the "12" from the last octet). Has anyone encountered this issue before? Andreas

4 12

help me about siremis.
by 문종민 25 Jan '13

25 Jan '13

I'm developers who trying to install siremis. will go in there and install siremis website. However, server-services -> domain list by pressing will get the following error. What can I do to fix this?

2 1

problem with db access
by MASebastian 25 Jan '13

25 Jan '13

Hi everyone, Recently I work on kamailio's snmpstats module. I'm trying to retrieve some data from database, the code pass the compilation but when starting kamailio always get this error. Jan 25 10:57:05 localhost kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/local/lib/kamailio/modules_k/snmpstats.so: undefined symbol: db_bind_mod I take the code from http://www.asipto.com/pub/kamailio-devel-guide/#c09database and ursloc module for example. not sure if I still miss something to make db access work. Any suggestion? Thanks, Sebastian

2 1

Websockets WSS problem with NOTIFY
by Pete Kelly 24 Jan '13

24 Jan '13

Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying "Compressed bit must be 0 if no negotiated deflate-frame extension" The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-s… Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point.

3 14

sipcapture + db_postgres gives lots of warnings on PostgreSQL server
by Øyvind Kolbu 24 Jan '13

24 Jan '13

Hi, while trying to use sipcapture together with PostgreSQL I've found that it stores the "\r\n" in SIP messages as "\015\012", though not without a lot of noise in the PostgreSQL log files. For every line inserted in PostgreSQL I get: [0] WARNING: nonstandard use of \\ in a string literal at character 763 [0] HINT: Use the escape string syntax for backslashes, e.g., E'\\'. [0] STATEMENT: insert into sip_capture (date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,msg) values ('2013-01-24 15:27:43',1359037663218925,'OPTIONS','','sip:129.240.253.134','','keepalive','533cb9e91f4b999cf76861cbb9ed54ed-149d','','','','','','5001ed5d-6979(a)127.0.0.1','','SIP/2.0/UDP 129.240.253.197;branch=z9hG4bKae6f.d4a02ac7.0','z9hG4bKae6f.d4a02ac7.0','10 OPTIONS','','','','kamailio (3.3.3 (i386/linux))','129.240.253.197',5060,'129.240.253.134',5060,'',0,'',0,1,2,'',1,'voipt1:21','OPTIONS sip:129.240.253.134 SIP/2.0\\015\\012Via: SIP/2.0/UDP 129.240.253.197;branch=z9hG4bKae6f.d4a02ac7.0\\015\\012To: sip:129.240.253.134\\015\\012From: sip:keepalive@edgetest.voip.uio.no;tag=533cb9e91f4b999cf76861cbb9ed54ed-149d\\015\\012CSeq: 10 OPTIONS\\015\\012Call-ID: 5001ed5d-6979(a)127.0.0.1\\015\\012Content-Length: 0\\015\\012User-Agent: kamailio (3.3.3 (i386/linux))\\015\\012\\015\\012') Kamailio as of latest git on the 3.3 branc + 3e580e1ee24f9b9ec4b1fa151b23ebc84f3db1cb. Same warnings when linked with both PostgreSQL 8.4.13 and 9.1.1. According to Alexandr Dubovikov MySQL stores '\r\n' as '\r\n'. -- Øyvind Kolbu

1 0

Decode contact with user paramer
by Gareth Rylance 24 Jan '13

24 Jan '13

When using the siputils function encode_contact() on a message that contains a user parameter e.g. <sip:XXXXX00015;phone-context=national@10.YYY.1.92;user=phone>;tag=88930 it encodes fine e.g. <sip:natted_ua*sip:XXXXX00015;phone-context=national**192.168.242.102*5080*udp@10.YYY.70.100 > but decode_contact() errors due to the ";" (there is an explicit check for it in the code) I am currently working round this using $rU = $(rU{s.replace,;,:}); to remove the ; as the user part is not needed for my application. Would it be safe to remove the check for ";" ? Gareth

3 6

Kamailio Multi tenant as registrar server ony
by andrea mucci 24 Jan '13

24 Jan '13

Hi Gurus, i have a basic question. i have a SIP PBX server that don't have the Multi Tenant Feature or Multi Company Feature. i now that Kamailio support this feature, so is possible to use the kamailio as registrar server and redirect all the services, SUBSCRIBE, NOTIFY, and RTP to the SIP PBX with corrects headers? Thanks for your help

3 2

Asterisk and dispatcher
by Ian French 24 Jan '13

24 Jan '13

> > Hi, > I've been working my way through this tutorial ( > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb) > and all it working well thanks. In the other benifits section you mention > that this configuration can be used with multiple instances of Asterisk. > I've configured my [TOASTERISK] route route[TOASTERISK] { ds_select_dst("1","4"); $sht(forw=>$ft)=$du; sl_send_reply("100","Trying"); xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk Cluster [$rm:$au].\n"); route(RELAY); exit; How can i achieve the same for my [FROMASTERISK] route as now with asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" and route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To [$rm:$au].\n"); return 1; return -1; } This allows calls only from cfg_get.asterisk.bindip (single gateway) where I would like gateways in the dispatcher list to be selected and marked as a positive match if(ds_is_from_list("1")) $sht(forw=>$ft)=$si; xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To [$rm:$au].\n"); return 1; return -1; } Can you help? Thanks in advance Ian

2 2

Guidelines to upgrade to 4.0.0
by Daniel-Constantin Mierla 23 Jan '13

23 Jan '13

Hello, the wiki page collecting the guidelines for upgrading from 3.3.x to 4.0.0 is available at: - http://www.kamailio.org/wiki/install/upgrade/3.3.x-to-4.0.0 I just added the SQL statements needed to upgrade the MySQL database, but there is lot of other stuff to be added. Hopefully, anyone noticing changes will add few notes and instructions about how to upgrade. The wiki can be edited if you make an account, it is not required to be a developer to add notes to this page. Would be great if users of postgres will add the sql statements specific for this database type. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Re: [SR-Users] sr-users Digest, Vol 92, Issue 50:Problem with 4.0 nightlies and IMS - sem_post undefined (Barry Flanagan)
by madhumanjusha＠integramicro.com 23 Jan '13

23 Jan '13

On Tue, January 22, 2013 1:58 pm, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Modify IP address in SDP (Austin Einter) > 2. Re: Modify IP address in SDP (Olle E. Johansson) > 3. Problem with 4.0 nightlies and IMS - sem_post undefined > (Barry Flanagan) > 4. Patch to resolve segfault in json module (Richard Brady) > 5. Re: Modify IP address in SDP (Austin Einter) > 6. Re: Patch to resolve segfault in json module > (Daniel-Constantin Mierla) > 7. Re: [sr-dev] FOSDEM (Daniel Pocock) > > > > ---------------------------------------------------------------------- > > > > > > > > > > > Message: 3 > Date: Tue, 22 Jan 2013 13:19:44 +0000 > From: Barry Flanagan <barry(a)flanagan.ie> > Subject: [SR-Users] Problem with 4.0 nightlies and IMS - sem_post > undefined To: sr-users(a)lists.sip-router.org > Message-ID: > <CALJb54-rPPbjgGR5iEKYztNGDvT11znthJzgMYovpp_t6N0S8w(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > > Hi > > > I am trying to run the Ubuntu nightlies from > http://deb.kamailio.org/kamailiodev-nightly and loading the p-cscf config > files. > > Things were working fine while I was on the 20130110 nightlies, but when > I > upgraded to 20130122 I get the following errors. Any have an idea why > sem_post is undefined and how to fix this? > > > > 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not > open module </usr/lib64/kamailio/modules/ims_registrar_pcscf.so>: > /usr/lib64/kamailio/modules/ims_registrar_pcscf.so: undefined symbol: > sem_post 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could > not open module </usr/lib64/kamailio/modules/cdp.so>: > /usr/lib64/kamailio/modules/cdp.so: undefined symbol: sem_post > 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not > open module </usr/lib64/kamailio/modules/ims_qos.so>: > /usr/lib64/kamailio/modules/ims_qos.so: undefined symbol: sem_post > > > Thanks > > > -Barry Flanagan >

2 1

Modify IP address in SDP
by Austin Einter 23 Jan '13

23 Jan '13

Dear All When 200 OK to INVITE passes through Kamailio proxy , I want to change the IP address in SDP to a specific IP address. Can this be done by just modifying configuration files or do I need to modify in source code. In case I need to modify source code, which files I need to look at. Kindly provide pointers for said problem. Thanks Austin

3 7

FOSDEM
by Peter Dunkley 22 Jan '13

22 Jan '13

Hello, Are there any plans for a Kamailio meal/get together on the Saturday evening this year? Peter

9 19

Patch to resolve segfault in json module
by Richard Brady 22 Jan '13

22 Jan '13

Hi folks If an empty or invalid JSON string is passed to the json_get_field function from the the json module, it causes a segfault. I have attached a patch. Regards, Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

2 1

Re: [SR-Users] Kamailio as a front-end for TLS-UDP
by Daniel-Constantin Mierla 22 Jan '13

22 Jan '13

Do no send replies to private address only, keep mailing list otherwise future such emails will be discarded. The contact has to stay the same, because it needs to be the address of the device. Record-Route and Path are designed to take care of routing back to lb, which will use tls to send to device. Cheers, Daniel On 1/22/13 12:26 PM, Anton Kvashenkin wrote: > Thanks for reply. Yes, kamailio acts as a balancer sending REGISTER > and INVITES, etc. The problem is that at L3-L4 it works well (sends > all datagram messages to 10.0.0.1:5060 <http://10.0.0.1:5060> UDP) but > at L7 the Contact header: <>;transport=tls. > > > 2013/1/22 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 1/22/13 10:56 AM, Anton Kvashenkin wrote: >> Guys, I've already asked @ G+ Kamailio coummunity but had no >> answer so I'm asking here: >> >> I'm trying to deploy +Kamailio >> <https://plus.google.com/114984922071338017631> as a front-end >> for TLS-UDP translation, as a back-end would be +FreeSWITCH >> <https://plus.google.com/104108080382960591162> . So, is kamailio >> so smart to fix Contact or Path header for ;transport=tls or it >> translates only on L3-L4 layer and I should fix those headers >> manually? > is it only for calls or you need to forward registrations to > kamailio as well? > > Everything is done automatically, for calls you just need to do > record routing. For registrations you have to use path module and > be sure that freeswitch supports sip path extension. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > -http://conference.kamailio.com - > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio as a front-end for TLS-UDP
by Anton Kvashenkin 22 Jan '13

22 Jan '13

Guys, I've already asked @ G+ Kamailio coummunity but had no answer so I'm asking here: I'm trying to deploy +Kamailio<https://plus.google.com/114984922071338017631> as a front-end for TLS-UDP translation, as a back-end would be +FreeSWITCH<https://plus.google.com/104108080382960591162> . So, is kamailio so smart to fix Contact or Path header for ;transport=tls or it translates only on L3-L4 layer and I should fix those headers manually?

2 1

VoIP diner at FOSDEM 2013
by Benjamin Henrion 22 Jan '13

22 Jan '13

Hi, I am organizing a diner for all Voice over IP hackers (Asterisk, Freeswitch, Kamailio, OpenSER, Sipvicious :-), etc…) during the Saturday evening of FOSDEM 2013. There is a complete track at the Telephony devroom as well: https://fosdem.org/2013/schedule/track/telephony/ More infos are registration: http://www.zoobab.com/voip-diner-fosdem Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

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Wiki page for what is new in v4.0.0
by Daniel-Constantin Mierla 22 Jan '13

22 Jan '13

Hello, there is a skeleton content for what is new in upcoming v4.0.0, but there are many new features missing in the lists. Please try to add short notes about the features you know they were added for 4.0. The wiki page is available at: * http://www.kamailio.org/wiki/features/new-in-devel Anyone that makes an account can edit: * https://www.kamailio.org/wiki/start?do=register Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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PATH advertised
by Olle E. Johansson 22 Jan '13

22 Jan '13

Hi! If I'm running Kamailio behind NAT and need to add a Path header with the outside IP address and port, that is configured as the "advertise" address - how do I do that? Can I add two Path headers at the same time? I can't find a way to add it like I can with Record-Route headers. /O

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Re: [SR-Users] sr-users Digest, Vol 92, Issue 43
by madhumanjusha＠integramicro.com 22 Jan '13

22 Jan '13

Thanks a lot for quick response. On Sat, January 19, 2013 3:59 am, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Mid call announcement(kamailio server) > (madhumanjusha(a)integramicro.com) > 2. Re: Mid call announcement(kamailio server) > (rabs(a)dimension-virtual.com) > 3. CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus > dlg - Kamailio 3.3.x (Rinor Hoxha) 4. Huge undertaking and in need of > advices from the community (Rumen Mihailov) > 5. Re: Huge undertaking and in need of advices from the > community (Stoyan Mihaylov) 6. Re: Invite - negotiate to asterisk as peer > not kamailio (Scott, Matt) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Fri, 18 Jan 2013 06:52:15 -0500 (EST) > From: madhumanjusha(a)integramicro.com > Subject: [SR-Users] Mid call announcement(kamailio server) > To: sr-users(a)lists.sip-router.org > Message-ID: > <39960.61.8.152.138.1358509935.squirrel(a)mail.integramicro.com> > Content-Type: text/plain;charset=iso-8859-1 > > > Hello, > As of now am working on Kamailio server. > My task is to announce in between call,my sip clients are EKIGA. > I should able to make my kamailio server act as controller and media > server,for that purpose my first step is : invite hold to called party. > second step is:: invite(sdp for .wav file) to caller and then am sending > rtp streaming.In wireshark am able to see rtp packets,but caller is not > playing the .wav file which am streaming through ortp library from > kamailio. > > Can anyone help me out with this problem?On top of it,is this possible in > reality? > > > Thanks & Regards, > Manjusha A. > Integra Micro Software Services (P) Ltd. > > > > > > > > ------------------------------ > > > Message: 2 > Date: Fri, 18 Jan 2013 15:14:49 +0100 (CET) > From: rabs(a)dimension-virtual.com > Subject: Re: [SR-Users] Mid call announcement(kamailio server) > To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <38001.212.40.242.42.1358518489.squirrel(a)www.rodriguezfeo.es> > Content-Type: text/plain;charset=iso-8859-1 > > >> Hello, >> As of now am working on Kamailio server. >> My task is to announce in between call,my sip clients are EKIGA. >> I should able to make my kamailio server act as controller and media >> server,for that purpose my first step is : invite hold to called party. >> second step is:: invite(sdp for .wav file) to caller and then am >> sending rtp streaming.In wireshark am able to see rtp packets,but caller >> is not playing the .wav file which am streaming through ortp library >> from kamailio. >> >> Can anyone help me out with this problem?On top of it,is this possible >> in reality? > > No, it's not possible using only kamailio, and thats because kamailio > it's a SIP proxy, it have nothing to do with media, for what you are > triying to get you need to router your calls throught a B2BUA, like sems > or asterisk. > > Best regards > > > > > > ------------------------------ > > > Message: 3 > Date: Fri, 18 Jan 2013 14:11:05 +0100 > From: Rinor Hoxha <rinorhoxha(a)gmail.com> > Subject: [SR-Users] CRITICAL: dialog [dlg_timer.c:205]: Trying to > update a bogus dlg - Kamailio 3.3.x To: sr-users(a)lists.sip-router.org > Message-ID: > <CAFvba4ipKqicMJ0qn9UNLXmhjGhf_83fzG=L5w2qVBpc7JC1GQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi list, > > > I'm having some issues with dialog module. From time to time I get the > following error: > > ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > [root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" | > sort | uniq -c > > 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil) > > This one was flooding my log file: tl=0x2aeeea1c6840 > > > 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy > /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy > /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this > happening and how to fix it. (I suspect that may be some issues with our > server memory...checking now...and will inform). However wanted to raise > this issue in case anyone else is having the same. > > Thanks, Rinor >

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Kamailio Development Workshop, Feb 14-15, 2013, Alicante, Spain
by Daniel-Constantin Mierla 21 Jan '13

21 Jan '13

Hello, I want to announce the next Kamailio Development Workshop, in Alicante, Spain, during Feb 14-15, 2013, organized by Asipto and University of Alicante, Department of Master Degrees in Telecommunication Engineering. You can find more details at: - http://www.kamailio.org/w/2013/01/kamailio-development-workshop-feb-14-15-2… Like the previous one of its kind (done in 2011 in Barcelona), this type of workshop targets to explain the source code structure and internal APIs of Kamailio, to speed up the learning of how to write own Kamailio extensions in C (e.g., write a module, which will be part also of the content). Its low registration fee is set to cover organizational logistics, the number of seats is limited, participants are accepted in the order of registration. Besides the workshop during the two days, on the evening of Feb 14 is planned a VoIP dinner. You can participate to it even if not coming to workshop. It will be nice to meet face to face and talk about latest trends in RTC/UC. If you need further details, contact me via email. Regards, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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all modules in one folder
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, please note that all modules are now back in one folder, respectively 'modules/'. There were few modules I couldn't compile due to missing exotic libraries, but based on grep, they should not depend on older modules paths. Anyhow, testing that the set of modules you are using compile and load file would be recommended. The folders modules_k and modules_s are now empty, with a placeholder .gitkeep, to be removed later if there is no need to revert any module relocation. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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updates to perl modules
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, there were several changes pushed to master branch related to perl modules: - perl was renamed to app_perl - to follow the naming pattern for embedded interpreters - embedded perl library is named now Kamailio (it was OpenSER so far, thanks to patches from Stoyan Mihaylov) - perlvdb was renamed to db_perlvd because it is a db driver module If you are using these modules, you will have to update your configuration file and perl scripts. Testing and feedback is appreciated. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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renaming to kamailio in perl and snmpstats modules
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, I'm writing to see if anyone has some spare time to take care of renaming from openser to kamailio in perl and snmpstats modules. I assume is more like search and replace, but has to be carefully done to be coherent and not break stuff there, so it might be a bit more than a sed command. The IANA enterprise number in MIBs is already the one for kamailio.org Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Testdrive the coming release 4.0 now!
by Olle E. Johansson 19 Jan '13

19 Jan '13

http://www.kamailio.org/w/2013/01/test-4-0-now/ Friends, We do need your help testing 4.0 now. - Compile on your systems, from small embedded systems to large servers, from Linux to *BSD and Solaris - Use with your configuration script - Test the new modules and functions - Read the updated documentation files On the link above, you have instructions on how to build from GIT sources and install a test system. We do look forward to your feedback! Let's all work to make this a very, very good and succesful release! Regards, /O ---- Kamailio on Twitter: @kamailioproject (also on Facebook and Google plus)

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Huge undertaking and in need of advices from the community
by Rumen Mihailov 18 Jan '13

18 Jan '13

Hi all, Thank you for all the help till now. Until now I was trying to setup completely redundant setup with Kamailio + asterisk boxes and for the moment i have the following configuration working. 2x Kamailios (Active/Passive) 2x Asterisks Kamailio is load balancing I can add more boxes if needed MySQL Cluster for all database needs Now the time has come and I will need to made this setup available for around 10K users. I will need to become an ITSP for an Internet Provider that is missing the telephony part from his package, so I have a couple of questions. 1. Is this setup OK to cover the needs of the customer base ? hardware will be decent 2. The end users will most probably need ATA adapters. What is the best cost effective solution I can go for ? 3. Do I need any other piece of software apart from the SIP Proxy and asterisk as I need the cost of implementation to be as low as possible. 4. Pricing...I can see that there are a lot of plans of the competitors that include 1000 minutes for free...Do you have any statistics what is the avarage % usage of prepaid minutes ? 5. Are there any other "stones" that I might hit and should be aware of ? Thank you for the information Best regards, Rumen

2 1

Merry Christmas!
by Daniel-Constantin Mierla 18 Jan '13

18 Jan '13

Another year getting to its end! Looking back, looks like one with the greatest achievements in the development of the project so far. Year is not done, so I save that summary for one week later, there is still stuff on its way to our source code repository. Now I want to thank to everyone promoting and contributing to the project, from developers to community members, and wish Merry Christmas and great winter holidays to all supporters of our project! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Mid call announcement(kamailio server)
by madhumanjusha＠integramicro.com 18 Jan '13

18 Jan '13

Hello, As of now am working on Kamailio server. My task is to announce in between call,my sip clients are EKIGA. I should able to make my kamailio server act as controller and media server,for that purpose my first step is : invite hold to called party. second step is:: invite(sdp for .wav file) to caller and then am sending rtp streaming.In wireshark am able to see rtp packets,but caller is not playing the .wav file which am streaming through ortp library from kamailio. Can anyone help me out with this problem?On top of it,is this possible in reality? Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

2 1

HEAD: Problems with $$ in xlog
by Klaus Darilion 17 Jan '13

17 Jan '13

Hi! In xlog I used to use $$ to escape the dollar sign to not be interpreted as PV, eg: xlog("L_INFO"," $$ru\n"); Current HEAD gives an error: ERROR: <core> [pvapi.c:757]: error searching pvar "" ERROR: <core> [pvapi.c:961]: wrong char [$/36] in [$] at [0 (0)] ERROR: xlog [xlog.c:363]: wrong format[ $$ru#012] ERROR: <core> [route.c:1214]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:627 Where there some changes? Thanks Klaus

2 2

Simple command to insert attributes to domain_attrs
by Philippe Sultan 17 Jan '13

17 Jan '13

Hi, I'm migrating from sip-router/Kamailio 3.1, SER flavour to the latest dev version, with MySQL support. I need to add attributes to domains, and therefore use the domain_attrs table. A handy command I used for that was ser_attr : ser_attr add domain=anydomain.voip attr=value Is there something similar I can use under the Kamailio flavour? kamctl does not seem to help here, unless I missed an option. Regards, Philippe

3 5

Users can't authenticate via external IP
by Rumen Mihailov 17 Jan '13

17 Jan '13

Hello Kamailio guys, I have the following setup Internet -----> REAL STATIC IP -----> Kamailio Virtual IP(10.2.45.101) I have setup everything and when I try to connect to kamailio from the inside I can successfully register. When I try to register to the REAL STATIC IP which is forwarded to 10.2.45.101 I get unauthorized. I have narrowed down the problem to this: auth_check("$fd", "sip", "1") as I said I can authenticate from the internal network but not from the Internet. What do I need to alter ? Side problem... why when using xlog("L_WARN","$Cxx TEST") kamailio fails to start ? seems that my version does not support the colored output... Thank you for your help guys...in advance. Best regards, Rumen http://itradebg.com

2 1

Server sip.iptel.org did not respond to register
by Faisal Farooq 17 Jan '13

17 Jan '13

Dear Support, We are using IPTEL SIP services. Now we are getting Error message on our VOIP software is " Server sip.iptel.org did not respond to register". Moreover, we can successfully ping and complete trace route the sip.iptel.org from our server but failed to register through sip.iptel.org server. Kindly suggest what is the cause and look into this matter. I am sending you our IPTEL SIP ID/Username and please find below. (1) 196211 (2) 195404 Regards, Faisal

2 1

i cannot log on to iptel.org
by kelly randolph 17 Jan '13

17 Jan '13

i cannot log on to iptel.org is everything okay???? thanx, kellyr

2 1

cant login
by Pasindu Hettiachchi 17 Jan '13

17 Jan '13

hi , my linksys router failed to register with iptel.org this happened from last night, any think that I have to change ? Pasindu

2 1

Error while loading modules, if given a custom library path
by Ajay Sabat 16 Jan '13

16 Jan '13

Hi I have trouble loading modules from a custom lib directory. It works fine if the libraries are located under PREFIX/lib/Kamailio/ (PREFIX is the path that I gave while building Kamailio). But on the target I will have to keep the libraries in a custom directory. I tried by changing the mpath in kamailio.cfg. Also I tried by giving the path in -L option while invoking kamailio. It is not working. It fails with the following errors, even if the libraries are present at the location where it compains. I searched but couldn't find any answer why it is failing, though it is supposed to work. Please help me. $ ./sbin/kamailio -E -d -f /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg -c loading modules under /home/ajay/projects/video_conf/lib/kamailio/:/home/ajay/projects/video_conf/lib/kamailio/modules/:/home/ajay/projects/video_conf/lib/kamailio/modules_k/ 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 222, column 12-23: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/kex.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 223, column 12-19: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/tmx.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 225, column 12-19: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules/sl.so>: libkcore.so.1: cannot open shared object file: No such file or directory $ ls -l /home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so -rwxr-xr-x 1 ajay ajay 155651 Jan 16 09:58 /home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so $ ls -l /home/ajay/projects/video_conf/lib/kamailio/libkmi.so* lrwxrwxrwx 1 ajay ajay 13 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so -> libkmi.so.1.0 lrwxrwxrwx 1 ajay ajay 13 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so.1 -> libkmi.so.1.0 -rwxr-xr-x 1 ajay ajay 78111 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so.1.0 Regards, Ajay

1 1

make option to build a lean kamailio image
by Ajay Sabat 16 Jan '13

16 Jan '13

Hi, I need to build a simple Kamailio server that could do just registration and basic call routing. I don't need database support, it is fine if it loses data during reboot. The only other option I may need is NAT support. So far I was building the server using make all as given below: $ make FLAVOUR=kamailio PREFIX="$HOME/projects/video_conf/sipserver" cfg $make all $make install I noticed that the total size of library and binary files is huge, little less than 30MB. I have to port and test the server on a target where I don't have much space. Can somebody please suggest me what make option would be appropriate for me? Is it fine to not build the ser modules under modules_s/ and what is make option to do so? I tried "make" instead of "make all", but it somehow builds everything during the step "make install" and the final result is same. Thanks, Ajay

2 2

TLS questions
by Fabian Borot 15 Jan '13

15 Jan '13

Hi, we need to implement TLS in our current setup and have some questions about it: 1- We have a farm of Asterisk servers behind our kamailio proxy. The call flow is Customers/Providers <--> Proxy <--> Asterisks. Is it possible to have TLS only between the Customers/Providers and the Proxy and then just regular SIP over UDP between the Proxy and the Asterisks? 2- Can we have TLS over both TCP and UDP at the same time? Thank you a lot fborot

3 3

Dokuwiki upgraded
by Daniel-Constantin Mierla 15 Jan '13

15 Jan '13

Hello, fyi, I upgraded dokuwiki to latest version, switching to the new default template (which is not bad at all, btw) as the old one is not yet fully compatible. If you find any issue, report to devel mailing list. It is about the wiki for v3.2+: - http://www.kamailio.org/wiki/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 2

NAPTR for TLS only
by Daniel Pocock 14 Jan '13

14 Jan '13

I'm just wondering if anyone can comment on expected and actual behavior if there is only a NAPTR record for TLS, e.g. I have: sip5060.net. IN NAPTR 10 0 "s" "SIPS+D2T" "" _sips._tcp.sip5060.net. and I don't have any entry for "SIP+D2U" or "SIP+D2T" If some third party Kamailio instance (e.g. sip-server.example.org) receives a request from a user trying to call sip:user@sip5060.net, with a sip: rather than sips: URI, should it (and will it) use the "SIPS+D2T" result, if no other result is available? Or would it ignore the NAPTR record and try to find the default SRV record such as _sip._udp.sip5060.net ? Should there be another NAPTR record to translate sip: to sips: using a regex perhaps, or would such a NAPTR be a bad thing?

3 4

Kamailio, Asterisk with TLS+SRTP
by Roberto Fichera 14 Jan '13

14 Jan '13

Hi All, I would setup a configuration where Kamailio authenticate asterisk SIP trunk using TLS and SRTP. At moment I was able to configure everything, including RTTProxy since most of the asterisks v1.8.19.1 are behind NAT. So far so good it works pretty good using standard authentication and the call goes straight between asterisks. But as soon as I move my configuration for both kamailio & asterisk to TLS+SRTP I'm not able to authenticate asterisk SIP trunks. Especially asterisk seems insisting to use the port 5060 even if I requested the TLS on 5061. kamailio v3.3.3 tls.cfg is configured as: [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/server.key certificate = /etc/pki/tls/certs/server.pem ca_list = /etc/pki/tls/certs/ca-bundle.crt #crl = //etc/kamailio/crl.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 = no require_certificate = no So my asterisk conf is the following: [general] tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/5002.pem tlscafile=/etc/asterisk/ca-bundle.crt tlscipher=ALL tlsclientmethod=tlsv1 tlsdontverifyserver=yes transport=tls,udp .... ..... and the SIP trunk is configured as [kamailio] type=peer insecure=invite,port nat=yes disallow=all allow=ulaw host=kamailio_ip outboundproxy=tls://kamailio_ip port=5061 defaultuser=5002 fromuser = 5002 fromdomain =mydomain secret=5002 qualify=yes dtmfmode=rfc2833 context=default callbackextension=5002 directmedia=nonat sendrpid=yes transport=tls encryption=yes register => tls://5002:5002@kamailio_ip:5061/5002 I still get error like: Jan 14 10:45:12] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:12] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:45:32] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:32] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:45:52] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:52] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:46:07] ERROR[7041]: tcptls.c:444 ast_tcptls_client_start: Unable to connect SIP socket to kamailio_ip:5060: Connection timed out Does anyone can suggest me something to read, try, check? Best regards. Roberto Fichera.

2 2

How to correctly check if the variable is set?
by Mino Haluz 14 Jan '13

14 Jan '13

Hi, how should I check if the value is set? if ($avp(s:test) == "") { or is there any null keyword ? If so, does it work for $avp, $sht, $var and $shv ? Thanks, Mino

3 4

Development is frozen for v4.0.0
by Daniel-Constantin Mierla 11 Jan '13

11 Jan '13

Hello, last midnight was the term to freeze the development for next major release v4.0.0. For a while, the GIT must branch should get only commits related to bug fixes, documentation improvements and updates of the additional tools. If any developer needs to store code of new features in the repository, use personal branches (or temporary branches in tmp/ when many developers work on the same code). After unfreeze, these branches can be merged in the master branch. For 1-2 months from now on, the main focus with the master branch is testing. The v4.0.0 will be out after mid of February, exact date to be set later based on the results during testing phase. There were many additions, so any kind of help testing is very appreciated. Fortunately the core and most of the very used classic modules were not changed much, ensuring good stability. But there are old very used pieces such as usrloc/registrar that got touched as well, so try to test as much as possible the components you are using in your configs. Provide any kind of feedback on sr-dev mailing list or the bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

PV: PAI vs. PPI
by Klaus Darilion 11 Jan '13

11 Jan '13

Hi! I just found out that there is only $ai for the whole P-Asserted-Identity URI, but lots of PVs for P-Preferred-URI: uri, domainpart, userpart ... Is there a reason for that, or is it just that nobody implemented it for PAI? I think it should be easy to extend it for PAI too. regards Klaus

2 1

Kamailio World Conference - Registration is open
by Daniel-Constantin Mierla 11 Jan '13

11 Jan '13

Hello, the registration for Kamailio World Conference is now open! You can see more details and register at: - http://conference.kamailio.com/k01/registration/ There is already a great group of speakers and interesting proposed talks. More regarding the content will be published in the near future, keep an eye on event's web site: - http://conference.kamailio.com If you are considering to speak at the conference, submit your proposal as soon as possible, the slots are filling up quickly. Looking forward to meeting many of you at the conference! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 3

Kamailio on Twitter
by Olle E. Johansson 11 Jan '13

11 Jan '13

Hello! For those of you that have missed it: You can now follow the project on Twitter: @kamailioproject https://twitter.com/kamailioproject Amongst the latest news are Kamailio at Fosdem 2013, Code freeze for kamailio 4.0 and the Kamailio World conference now open for registrations. Tonight we went into code freeze for the coming release of Kamailio 4.0. This means that developers can not add any more features to this release, only fix bugs. In order to fix bugs we will need your help testing. After the weekend we'll come back with instructions on how to test and report bugs. The information is already available on the web site. Have a great weekend! /O

1 0

LCR
by Douglas Ugalde 11 Jan '13

11 Jan '13

Hi, Im trying to configure LCR in Kamailio 3.3.3 but I dont Know how can I do to fix this error: ERROR: lcr [lcr_mod.c:1840]: error while parsing R-URI This is my LCR configuration block: route[LCR] { if(!load_gws(1)){ sl_send_reply("500", "Internal server error, unable to load gateways"); xlog("L_NOTICE","Internal server error, unable to load gateways"); break; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); break; } } I not sure if this configuration thats ok, please somebody help me. Note: Params and modules are already loaded, sorry for my english. Best regards.

2 1

Problem with MSILO module forwarding messages
by Owen Lynch 10 Jan '13

10 Jan '13

Hi, I have the kamailio 3.3.2 msilo module basically working but I cannot configure it to forward stored messages to the registered UA. Instead it forwards to itself (via DNS lookup on the domain name). This is causing me grief because I am getting recursive message handling. The module overview describes this behaviour: "Every time when a user registers with Kamailio, the module is looking in database for offline messages intended for that user. All of them will be sent to contact address provided in REGISTER request." The REGISTER request contact header looks OK: Contact: <sip:hxqecvtn@192.168.2.236:57532 ;transport=tls>;expires=3600;received="sip:192.168.2.236:53628 ;transport=TLS";+sip.instance="<urn:uuid:1ee63413-3569-46dc-a9ee-34c6e4e1403c>" However the 'outbound_proxy' parameter seems to contradict this: "The SIP address used as next hop when sending the message. Very useful when using Kamailio with a domain name not in DNS, or when using a separate Kamailio instance for msilo processing. If not set, the message will be sent to the address in destination URI." So, when 'outbound_proxy' is not set, the message is sent to the address in the destination URI, rather than the UA contact address? Can the module be configured to forward stored messages directly to the UA contact address? Thanks, Owen Lynch

2 1

Socket error
by Douglas Ugalde 10 Jan '13

10 Jan '13

Hi, Please I need help with this error: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) I'm using Kamailio 3.3.3 I call 2 times before got this error, how can I do to resolved it, I really really appreciated any help. I attached my Kamailio.cfg. Really sorry for my English. Best Regards.

2 1

NAT Problem! Not relay ACK,CANCEL and BYE to other UA
by Nguyen Anh Tuan 10 Jan '13

10 Jan '13

Hi All, I have 2 interfaces - External A (behind NAT) and Internal B (LAN) in Kamailio Load Balance, and enable Multi-Homing function "mhomed=1". Load Balance dispatches to Kamailio SIP Servers via the Internal B (LAN) (each Kamailio SIP server has both proxy and registrar in one box). For RTP relay, I use RTPproxy in other server, has 2 interfaces A&B like Load Balance. If using 2 interfaces in Load Balance, Registration and calling NORMAL with 2 users behind NAT, but the call will be interrupted after 30 seconds due to the ACK message (after Ringing, 200 OK) STOP in Load Balance in return and not relayed to other UA . CANCEL and BYE have same error like ACK. I tried using only one interface for load balancing (External = Internal IP), and SIP Server binds to Load Balance in the same network, EVERYTHING IS OK. ACK, CANCEL and BYE are not missing behind NAT. For NAT Traversal I use PATH Module in Load Balance, and add "Received" like this: ... modparam("path", "use_received", 1) .. .. record_route(); add_path_received(); force_rport(); t_relay; .. .. And in SIP Server enable Path Parameter for Registrar: modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 1) modparam("registrar", "path_use_received", 1) Enable "#!define WITH_NAT" in default kamailio.cfg for RTP Controlling, but I removed all fix_nated_register and fix_nated_contact in SIP Server, because using PATH module.I try enable both fix_nat_register and fix_nated_contact, or only fix_nat_register but NOT SOLVE Problem "missing ACK,CANCEL, and BYE in Load Balance" behind NAT. Best Regard, Tuan.

2 1

Testing SIP Outbound / RFC5626 (Was: Freezing for next major release)
by Richard Brady 10 Jan '13

10 Jan '13

Hi Peter Great work on this! We'd like to help you test. The test I have in mind, which we could create using SIPp, would be to register multiple contacts with the same instance-id (i.e. "sip.instance" param) but different reg-id params. Then send an INVITE to that AoR and make sure the forking is only per instance-id and not per reg-id. This could be repeated in multiple permutations of instance-ids and reg-ids. This would be a test of save() and lookup() more than anything else. Is that what you had in mind? Richard On 3 January 2013 14:13, Peter Dunkley <peter.dunkley(a)crocodile-rcs.com>wrote: > I hope to get the outbound edge proxy and flow timer stuff into master by > Monday, but it could really do with some additional testing especially in > conjunction with the reg-id stuff in registrar/usrloc (which I have no idea > how to use). > > Peter > >

5 14

LCR with Siremis
by Douglas Ugalde 09 Jan '13

09 Jan '13

Hi, I was configuring the modparams for LCR and Dialplan, I taken from kamailio web site, but I think the params are outdated, because we need to erase some lines to match with the tables in the DB, Im using Kamailio 3.3.3 and Siremis 3.3 Please can you give 1 example about how to make outbounds calls with siremis 3.3 (send calls to gateway)? I already have the lcr and dialplan module loaded with params. Where can I get update documentation about Siremis 3.3 and Kamailio 3.3.3. Thanks for your help.

1 0

Freezing development for v4.0.0
by Daniel-Constantin Mierla 09 Jan '13

09 Jan '13

Hello, a (26-hours before) reminder that development for next major release v4.0.0 will be frozen tomorrow, Thursday, January 10, 2013, at 23.59 GMT. If any developer has new features to push in the git master branch, please do it before. After that, for a while, the master should get only bug fixes and improvements to documentation and addition tools (e.g., bringing kamctl up to date). New features have to be kept in personal branches until the master branch is open again. The focus should be on testing from the day after tomorrow on, I hope the users community will help us here as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Having "modparam" options in the database
by Glenn O. Larsen 09 Jan '13

09 Jan '13

On Tue, Jan 8, 2013 at 9:21 AM, Olle E. Johansson <oej at edvina.net> wrote: >>>> Is there a way to read modparam config from a database? > In most cases, there should be no need to change modparams at > runtime. > > Can you tell us more about which parameter you want to change during > runtime that doesn't have a mI/rpc reload command? Hi Olle! The module we have been using, and wanted database configuration for was the pike module. But and after spending some more time looking at alternative modules, I see that we can use other modules to help us with traffic control/blocking, like the pipelimit and the ratelimit (both these modules have some sort of database integration, need to look more into this). I was so focused on the pike module, that I didn't see these other modules.. :) -- mvh Glenn O Larsen

3 3

Core fm_realloc function
by Bruno Bresciani 08 Jan '13

08 Jan '13

Hi all, I observed two at 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536. I don't know exactly what may be caused this core because the debug of kamailio is disabled, but looking at backtrace (gdb) and source code I could notice a relationship with TLS connection establishment (handshake). Other important fact is that core was generated by the kamalio itself with signal 6 (abort) because it can't find the pointer to memory realloc. I'd like to know if someone already observed this issue and if my analyse above is correct? Bellow is backtrace full with gdb: Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 6, Aborted. #0 0x00f30402 in __kernel_vsyscall () (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 0x00f30402 in __kernel_vsyscall () No symbol table info available. #1 0x009fec10 in raise () from /lib/libc.so.6 No symbol table info available. #2 0x00a00521 in abort () from /lib/libc.so.6 No symbol table info available. #3 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536 f = (struct fm_frag *) 0xb61f0bb8 pf = <value optimized out> orig_size = 18 L=ÃbÂ¬Ã2n = (struct fm_frag *) 0xb61f0bd2 ptr = <value optimized out> hash = 23548 #4 0x0051a87f in ser_realloc (ptr=0xb61f0bc0, size=32) at ../../mem/shm_mem.h:266 No locals. #5 0x00460295 in OPENSSL_cleanse () from /lib/libcrypto.so.6 No symbol table info available. #6 0x004609f7 in CRYPTO_realloc () from /lib/libcrypto.so.6 No symbol table info available. #7 0x003efb22 in sk_insert () from /lib/libcrypto.so.6 No symbol table info available. #8 0x003efbaa in sk_push () from /lib/libcrypto.so.6 No symbol table info available. #9 0x007c2ac3 in ssl_bytes_to_cipher_list () from /lib/libssl.so.6 No symbol table info available. #10 0x007aa9f3 in ssl3_get_client_hello () from /lib/libssl.so.6 No symbol table info available. #11 0x007ab815 in ssl3_accept () from /lib/libssl.so.6 No symbol table info available. #12 0x007c1c3a in SSL_accept () from /lib/libssl.so.6 No symbol table info available. #13 0x005278e3 in tls_accept (c=0xb621af88, error=0xbf91dfc8) at tls_server.c:346 ret = <value optimized out> ssl = (SSL *) 0xb61c1878 cert = <value optimized out> tls_c = (struct tls_extra_data *) 0xb6208e58 tls_log = <value optimized out> #14 0x0052da8e in tls_read_f (c=0xb621af88, flags=0xbf91e200) at tls_server.c:1028 r = (struct tcp_req *) 0xb621affc bytes_free = 4095 bytes_read = 70 read_size = 4095 ssl_error = 0 ssl_read = 0 ssl = (SSL *) 0xb61c1878 rd_buf = "\026\003\001\000A\001\000\000=\003\001OÃ¿\vÂ¡ywP\fkÃDÃ 0/\000\a\000\005\001\000ÃÃÃÃ¯000\000\026\0009\0008\0005\000\026\000\023\000\n\0003\0002\00 202i<Ã¤<Ã¥Ã¾\031Ã¬234Â \237Ãª030Y){lÑ¶"...06Ã'Ãr9Ã¬Ã«Â¸{[Â©\t\205\037\036Ã¿HPH5Âµ\t(Ã©32Ã027Ã¦Ã²036Â¾\"\rÔ¤YÃ215uÂº\027)Ã¶000Ã±215\vÃÃ Ã©Ea\023xÃ©J\206T\204CÃ wr_buf = "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"... rd = {buf = 0xbf90dfa4 "\026\003\001", pos = 70, used = 70, size = 65536} wr = { buf = 0xbf8fdfa4 "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"..., pos = 0, used = 0, size = 65536} tls_c = (struct tls_extra_data *) 0xb6208e58 enc_rd_buf = (struct tls_rd_buf *) 0x0 n = 0 flush_flags = <value optimized out> err_src = 0x541a83 "TLS read:" x = <value optimized out> #15 0x08158ade in tcp_read_headers (c=0xb621af88, read_flags=0xbf91e200) at tcp_read.c:406 bytes = -1 remaining = <value optimized out> p = <value optimized out> r = (struct tcp_req *) 0xb621affc #16 0x08158fd4 in tcp_read_req (con=0xb621af88, bytes_read=0xbf91e204, read_flags=0xbf91e200) at tcp_read.c:871 bytes = <value optimized out> total_bytes = 0 resp = <value optimized out> size = <value optimized out> req = (struct tcp_req *) 0xb621affc dst = {send_sock = 0x1, to = {s = {sa_family = 57540, sa_data = "\221Â¿\020\000\000\000\000\000\000\000\020\000\000"}, sin = {sin_family = 57540, sin_port = 49041, sin_addr = {s_addr = 16}, sin_zero = "\000\000\000\000\020\000\000"}, sin6 = {sin6_family = 57540, sin6_port = 49041, sin6_flowinfo = 16, sin6_addr = {in6_u = {u6_addr8 = "\000\000\000\000\020\000\000\000\001\000\000\000\001\000\000", u6_addr16 = {0, 0, 16, 0, 1, 0, 1, 0}, u6_addr32 = {0, 16, 1, 1}}}, sin6_scope_id = 12}}, id = -1080958468, proto = 4 '\004', send_flags = {f = 0 '\0', blst_imask = 0 '\0'}} c = 10 '\n' #17 0x08159a0f in handle_io (fm=0x83296b0, events=1, idx=-1) at tcp_read.c:1150 ret = 4 n = 4 read_flags = 1 con = (struct tcp_connection *) 0xb621af88 s = 12 resp = <value optimized out> t = <value optimized out> #18 0x0815d653 in tcp_receive_loop (unix_sock=35) at io_wait.h:1091 No locals. ---Type <return> to continue, or q <return> to quit--- #19 0x0812885b in tcp_init_children () at tcp_main.c:4819 r = 3 reader_fd_1 = 35 pid = <value optimized out> si = <value optimized out> #20 0x080ad412 in main_loop () at main.c:1632 i = 4 pid = <value optimized out> si = (struct socket_info *) 0x0 si_desc = "udp receiver child=3 sock=192.168.166.31:5060 \000\221Â¿\002\000\000\000/\200\220\000\001\000\000\000\000\000H\000Â¨Ã¤21Â¿\001\000\000\000Â°Â¶Ã¨026Â¶\000\000\000\000\000ÃÂµ\030s2\b\002\000\000\000Ã¢f\b\000ÃÂµ\001\000\000\000\000\000\000\000\001\000\000\000\000\000\000\000Ã¿Ã¿Ã¿Ã¿Â¸Ã¤21Â¿" #21 0x080afbe7 in main (argc=3, argv=dwarf2_read_address: Corrupted DWARF expression. ) at main.c:2398 cfg_stream = (FILE *) 0x8a2d008 c = <value optimized out> r = 0 tmp = 0xbf91e674 "\212Ã¾\221Â¿\233Ã¾\221Â¿Â°Ã¾\221Â¿Â»Ã¾\221Â¿Ã\221Â¿Ã¹1Â¿\020Ã¿\221Â¿DÃ¿\221Â¿LÃ¿\221Â¿WÃ¿\221Â¿]Ã¿\221Â¿oÃ¿\221Â¿{Ã¿\221Â¿\202Ã¿\221 tmp_len = 10492229 port = 136192361 proto = -1080957480 ret = <value optimized out> seed = 21782655 rfd = 4 debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = <value optimized out> p = <value optimized out> Best Regards

4 24

ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES)
by Shreyansh Purwar 08 Jan '13

08 Jan '13

Hello Everyone, While creating an account on Kamailio Sip server (I am running the command "kamctl add test test1"), it is asking for password of MySQL Server and after entering the password, I am getting following errors :- - ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES) - ERROR: introducing the new user 'test' to the database failed Please help me in resolving this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

raw_query is crashing....
by Krishna Kurapati 08 Jan '13

08 Jan '13

Hi, Since there is no way to limit the result set to 1, I tried to use raw_query inside msilo module. Everytime it executes, it is crashing. The same query if I run in mysql shell, it is returning successfully. Here is the code That I added: query_len = snprintf(query, 2048, "SELECT * FROM %.*s WHERE %.*s='%.*s' AND %.*s='%.*s' AND %.*s!=%i AND %.*s<%i LIMIT 1;", ms_db_table.len, ms_db_table.s, sc_uri_touser.len, sc_uri_touser.s, puri.user.len, puri.user.s, sc_uri_tohost.len, sc_uri_tohost.s, puri.host.len, puri.host.s, sc_status.len, sc_status.s, 200, sc_status.len, sc_status.s, MSG_NOTIFY_OFFSET); if (query_len < 0) { LM_ERR("error in sql snprintf"); goto error; } LM_DBG("m_dump_msg: Query=%s\n", query); query_str.s = query; query_str.len = query_len; if (msilo_dbf.raw_query(db_con, &query_str, &db_res) < 0) { LM_ERR("Failed to query database. Query = %s\n", query); goto error; } The Debug Log at Level 4 shows following: 11(7520) DEBUG: msilo [msilo.c:1120]: m_dump_msg: Query=SELECT * FROM silo WHERE tousername='968257197' AND todomain='50.62.1.7' AND status!=200 AND status<1000 LIMIT 1; 11(7520) DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x7fdd4b605540 11(7520) DEBUG: db_mysql [km_res.c:68]: 16 columns returned from the query 11(7520) DEBUG: <core> [db_res.c:155]: allocate 128 bytes for result names at 0x7fdd4b5d8dc0 11(7520) DEBUG: <core> [db_res.c:165]: allocate 64 bytes for result types at 0x7fdd4b5d8e60 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x7fdd4b5d8d90 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d90)[0]=[id] 11(7520) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x7fdd4b5d8d60 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d60)[1]=[src_addr] 11(7520) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x7fdd4b5d8d30 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d30)[2]=[dst_addr] 11(7520) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type <skip> 11(7520) DEBUG: <core> [db_val.c:117]: converting STRING [ac3aeddd9b6f45e6822c824aa2c133b2] 11(7520) DEBUG: <core> [db_val.c:73]: converting INT [0] 11(7520) DEBUG: <core> [db_val.c:73]: converting INT [1] 26(7552) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 22 26(7552) DEBUG: <core> [tcp_main.c:3592]: DBG: handle_ser_child: dead child 11, pid 7520 (shutting down?) 26(7552) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x842d20, 22, -1, 0x0) fd_no=44 called 26(7552) DEBUG: <core> [tcp_main.c:3353]: DBG: handle_tcp_child: dead tcp child 1 (pid 7520, no 11) (shutting down?) 26(7552) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x842d20, 25, -1, 0x0) fd_no=43 called 0(7509) ALERT: <core> [main.c:787]: child process 7520 exited by a signal 11 0(7509) ALERT: <core> [main.c:790]: core was not generated 0(7509) INFO: <core> [main.c:802]: INFO: terminating due to SIGCHLD As you can see something happening to Database connection. Not sure why I am unable to use raw_query. Normal API query from msilo works just fine. What am I doing wrong to make it crash? Thanks Krish Kura

2 1

Having "modparam" options in the database
by Glenn O Larsen 08 Jan '13

08 Jan '13

Hi guys, I have tried to Google this for some time now, but can't seem to find an answer. Is there a way to read modparam config from a database? -- Best regards, Glenn Larsen

3 3

LCR weight
by Ricardo Martinez 08 Jan '13

08 Jan '13

Hello. I have a doubt about the WEIGHT value in the LCR module. In the past we used the Kamailio version 1.5.2 and in that version LCR weight was calculated by a little script called lcr_weight_script.php. There was not “percent” values like 50% and 50% for two gateways despite that the probabilities were numbers calculated by the script. My question is : For Kamailio version 3.2.4 do I have to use the same script to calculate the weight?? I was almost sure that the weight column was a probability value. For example if I have 3 gateways, with 50% 25% and 25% The weight column should be : 50, 25 and 25 ? or 45, 32 and 32 as the script indicates??? [root@vlcr utils]# ./lcr_weight_test.sh 45 32 32 weight 45 probability 0.5235 weight 32 probability 0.2352 weight 32 probability 0.2413 Please enlighten me here. Thanks Ricardo.-

2 1

TCP Path received quotes
by Spencer Thomason 07 Jan '13

07 Jan '13

Hello, I'm forwarding registrations to Freeswitch and adding a Path header using add_path_received() for endpoints behind NAT. When TCP is used, this results in a header like this: Path: <sip:a.a.a.a:5070;lr;received="sip:b.b.b.b:35430;transport=tcp"> Freeswitch doesn't like the transport protocol inside the quotations and refuses to route a call to the registered user. Constructing the header manually without the quotes such as: append_hf("Path: <sip:$Ri:$Rp;lr;received=sip:$si:$sp;transport=$pr>\r\n"); solves the problem. Is there a reason for the quotes? Best Regards, Spencer

4 7

Multi tenant Kamailio
by Rumen Mihailov 07 Jan '13

07 Jan '13

Hello Kamailio, VoIP and asterisk gurus, I have the following really complicated scenario to solve. I have CompanyA which has extensions 100-120 I have CompanyB which has extensions 100-115 I have CompanyC which has extensions 100-130 Now I want to have all those extensions on a single kamailio box, but I want to achieve the following. CompanyA calls 110 and this rings CompanyA 110 not CompanyB or CompanyC. What I want to achive is multitenancy. I know how to make this in asterisk but I want to take advantage of Kamailio's load balancing and HA features. Sammy from the mailing lists advised that I use multidomain and actually I saw that this module is indeed intended to do that, however I have no clue how to use it. I mean do I have to setup a domain for all the companies ? For instance companya.itradebg.com, companyb.itradebg.com and then advise the customers to register to their respective domain or what ? This concept is really a bit tough for me despite all the help from Sammy for which I am really greatful. I would greatly appreciate any tutorials or links to some useful documentation apart from the module documentation. Thank you in advance guys. Best regards, Rumen http://itradebg.com

2 2

NAT'ed RLS setup
by Dmytro Bogovych 07 Jan '13

07 Jan '13

Greetings. I have kamailio 3.3.3 setup on server behind the nat. Nat forwards ports for tls/tcp/udp. This setup worked without rls ok. I added necessary modules - rls, pua, xcap etc. Updated a config file and db. Client application uploads xcap documents ok; after app sends subscribe request to list address. Rls/pua modules generate subscriptions to entries in list. But dns resolver finds the public ip as target for subscription. The result is: sip packets cannot reach the server - they are sent to public ip of nat machine. I see it in ngrep output. Nat does not route them back. I add the domain name and _sip._udp.domain with internal ip address to /etc/hosts. But kamailio does not use /etc/hosts when dns caching is active. I do not want to loose dns caching. I made an attempt to run without caching but server was not stable. It crashed couple of times - so i stopped these attempts. Is there way to put eternal domain -> ip bindings into dns cache? I want to put bingings to internal ip - hope it will help. Maybe there is another way to handle this problem?

1 0

Timeout for t_uac_dlg per request
by David 07 Jan '13

07 Jan '13

Hello, I am running Kamailio 3.2.x, I am using udp mi to send a t_uac_dlg command, :t_uac_dlg: NOTIFY sip:phone1@proxy.lan . . "From:sip:daemon@proxy.lan To:<sip:phone1@proxy.lan> Contact:sip:daemon@proxy.lan" # I would like to know if I can change the timeout for this request. Running a packet sniffer on the proxy, I see the packet is retransmitted after 500ms, then 1000ms, then 2000ms, etc... This is the proper behaviour for SIP, so we can see my tm module is working well. My problem is that I want a response within 3 seconds ( I know this doesn't respect the normal SIP timer values ). I do not want to change the timers in the tm module for the entire proxy, so my question is if there is a way to set the timeout for the dialog created by the t_uac_dlg command, and can I set it on a per dialog basis ? David

1 0

TLS with self-singned certificates
by Iurii Andamasov 07 Jan '13

07 Jan '13

Hello, trying setup kamailio with TLS, have 2 peers, tls.cfg: http://pastebin.com/DvdDzx0v i'm getting Jan 4 10:38:43 fs-tls /usr/sbin/kamailio[3798]: ERROR: tls [tls_server.c:1190]: TLS read:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure Can someone point me to solution? Thanks

3 4

tls_domain.h:37: fatal error: openssl/ssl.h: No such file or directory
by Shreyansh Purwar 07 Jan '13

07 Jan '13

Hello Everyone, While building kamailio server 3.0.0 with TLS(I have removed tls from exclude module variable) I am getting following error :- - tls_domain.h:37: fatal error: openssl/ssl.h: No such file or directory Please help me how to resolve this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

Malicious traffic comming from 37.75.210.90
by Nick Khamis 06 Jan '13

06 Jan '13

Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use "972599779558" N.

1 0

dialog ka_timer and ka_interval problem
by Rinor Hoxha 06 Jan '13

06 Jan '13

Hi, I had an issue with dialog module of kamailio v3.3.3. Is the same described on http://lists.sip-router.org/pipermail/sr-users/2012-November/075735.html. I think the issue is related to dgl_hash.c on line:192 *dlg_ka_list_head = NULL; should be *dlg_ka_list_tail = NULL; and line:216 *dlg_ka_list_tail = dka; should be *dlg_ka_list_head = dka;. I will test it as soon as I can, but meanwhile I'm attaching the patch that should fix it so you can test it and report back the results if you want to. Br, Rinor

2 1

Kamailio DB version error
by Douglas Ugalde 06 Jan '13

06 Jan '13

Hi, I want to load the "dialplan.so" with parameters but when I reload kamailio, I got this error: /usr/local/sbin/kamailio[1348]: ERROR: <core> [db.c:439]: invalid version 0 for table my_table found, expected 1 (check table structure and table "version") Im using kamailio 3.3.3 Please what can I do to resolve it? Regards.

2 1

Next major release proposed as v4.0.0
by Daniel-Constantin Mierla 04 Jan '13

04 Jan '13

Hello, using v4.0.0 for the next major release was proposed and debated at previous development IRC meeting. First was about the addition of a new transport layer (websockets) along with other important new features. During the debate was touched the completion of kamailio-ser integration (no more duplicated modules) as another milestone for increasing the first digit (which was finished just before the end of last year). Now we also have the IMS modules, pushed in the GIT master branch few days ago. It seems that there are lot of good reasons for numbering the next major version 4.0.0. Other comments, opinions? Have a great and happy year ahead, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 4

Multiple rtimer processes + mqueue
by Alex Balashov 04 Jan '13

04 Jan '13

Will multiple rtimer processes consuming the same mqueue, i.e. via a blocking mechanism while(mq_fetch(...)) ... hurt anything? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

Freezing for next major release
by Daniel-Constantin Mierla 04 Jan '13

04 Jan '13

Hello, based on last irc devel meeting and feedback afterwards, most of the people seem to be fine releasing next major release a bit earlier, in order to: - make websockets support and the other new features available in a stable version - allow proper development time frame for other planned new features that wouldn't have been ready if the release would be 1-2 months later So, considering the winter holidays, I propose to freeze development on January 7, 2013, with full release about one month later. It seems that many people are already using devel version to play with websockets, meaning that we are staying very well with testing. Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 5

"maximum number of branches exceeded" after kamalilio restart
by Andrew Pogrebennyk 04 Jan '13

04 Jan '13

Hello, I'm getting some errors - if I restart kamailio on a live system, the first call after restart fails with "max nr of branches exceeded". However, we are not doing append_branch() or anything fancy for that call at all: Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: <core> [dset.c:306]: max nr of branches exceeded Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: registrar [lookup.c:298]: failed to append a branch Also it appears on the other system with a different config: Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:656]: ERROR: add_uac: maximum number of branches exceeded Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed The second call however goes through and this error does not appear until the next kamailio restart. What could be the problem here? P.S. The kamailio version is 3.3.2. Andrew

1 0

km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory
by Shreyansh Purwar 04 Jan '13

04 Jan '13

Hello Everyone, I need a help, I was installing kamailio SIP server according to the instruction provided at wiki page - http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git but while compiling it(make all) i am getting following error - km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory Can anybody tell me how to resolve this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

Re: [SR-Users] [sr-dev] Freezing for next major release
by Peter Dunkley 03 Jan '13

03 Jan '13

I hope to get the outbound edge proxy and flow timer stuff into master by Monday, but it could really do with some additional testing especially in conjunction with the reg-id stuff in registrar/usrloc (which I have no idea how to use). Peter Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >Hello, > >a reminder that the freezing date of the development for the next major >release is next Monday, January 7. > >Considering it was vacation time so far, we can prolong development for >few days if it is needed - just reply here. > >Besides few small additions, I want to add internal hash table to track >MSRP connections, hopefully I should be able to do it in time. If not, I >will ask for few more days as well. > >Cheers, >Daniel > >On 12/5/12 5:05 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> based on last irc devel meeting and feedback afterwards, most of the >> people seem to be fine releasing next major release a bit earlier, in >> order to: >> - make websockets support and the other new features available in a >> stable version >> - allow proper development time frame for other planned new features >> that wouldn't have been ready if the release would be 1-2 months later >> >> So, considering the winter holidays, I propose to freeze development >> on January 7, 2013, with full release about one month later. >> >> It seems that many people are already using devel version to play with >> websockets, meaning that we are staying very well with testing. >> >> Other opinions? >> >> Cheers, >> Daniel >> > >-- >Daniel-Constantin Mierla - http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > >_______________________________________________ >sr-dev mailing list >sr-dev(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

1 0

recreating kamilio_fifo without restart
by Uri Shacked 03 Jan '13

03 Jan '13

Hi, I entered the command "kamailio -v" instead of "kamailio -V". It looks like it starts kamailio again even that it was running.... the problem is that the kamctl_fifo and kamctl_ctl file are gone now. how do i create the files again without restarting kamailio? Thanks, Uri

3 4

All The Best And A Happy 2013!
by Daniel-Constantin Mierla 31 Dec '12

31 Dec '12

This was it for 2012, one of the greatest years in the history of the project! One major release was done in the summer (version 3.3.x), another one is few days before testing phase, most probably to be out by mid of February. The next major release includes several great features and edevelopments: - support for websockets - IMS extensions in the main GIT branch - no more duplicated modules, integration of Kamailio and SER ended There are big plans for 2013. First is the next major release, together with participation at FOSDEM conference in Berlin. A bunch of other events will open the path to our first dedicated conference for Kamailio project - Kamailio World. We hope to meet many of you at Kamailio World conference, in Berlin, Germany, during April 16-17. Last organizing bits were sorted out and registration will open in the first days of 2013 -- you can see more details at: - http://conference.kamailio.com Looking forward to a great 2013 for Kamailio and everyone involved in the project, from developers to community members! All the best! Happy New Year! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio TLS setup and certificate
by Burak ÖZTÜRK 31 Dec '12

31 Dec '12

Hi, I m a new kamailio user. Kamailio version is 3.3. I want to enable kamailio's tls support. I fallowed instructions from this link : http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates And I also enabled tls module in the kamailio.cfg file. (definig WITH_TLS and adding listen=tls:myIP:5061 command) But after trying to start kamailio i got "failed to start" error. When I checked the logs there is nothing about tls or ssl. Actually I got an error while fallowing the link above. "openssl s_client -connect localhost:5061 -tls1" command returns connection failed 111 error. I thought that this is the reason failed to start. But where or what am I doing wrong? I searched the web and hear about SRV records. But there is no explanations about this, on your guide/wiki. I also read this link too but cant get my answer : http://sip-router.org/docbook/sip-router/branch/master/modules/tls/tls.html Is there any idea or a better well-explained user guide about tls support? Regards, Burak

2 1

Zero left padding IP in Kamailio
by Techie Sup 30 Dec '12

30 Dec '12

Hello, My IP is 192.168.1.25 and I have added left zero padding i,e: 192.168.001.025 in the configuration file and try to start kamailio, it works.. but along with that rtpproxy does not start. Once I revert back it works. My script requires to add zero left padding for a specific purpose and for that I cannot remove the left zero padding. Any solution to get this work? Regards Selvam

2 1

kamailio-ser merging finished
by Daniel-Constantin Mierla 29 Dec '12

29 Dec '12

Hello, in November 2008 was decided to start integration of kamailio and ser project, to get to same source code. About one year later, there was a single source code tree, but there were several duplicated modules that kept some confusion around the two projects. As the time passed, it was shown better what modules are in the focus, making the decisions to sort them out easier. The past weeks were dedicated to removing duplicated, process that just ended -- at this moment there is no duplicated module, all of them being integrated. In summary, the modules from ser that had no conflict were moved as is to modules folder, eventually after rename. The modules with conflicts were merged, adding missing features to the most developed ones, moving the others to obsolete folder. It happened that kamailio version of the modules stayed in the majority of the cases, because they had larger set of features (the core in v3.0.0 was kept from ser). Please note that the extra features from ser modules might not be in the same form, but alternatives that should offer same functionality at the end. People using ser flavour so far should start testing the master branch. Do ask questions on the mailing lists if something that you used to have in config is not longer available, so we can assist in migration and, hopefully, create a migration guide on wiki. There might few bits and pieces to tune here and there, soon we will get into code freezing period for testing the next major release, a good time for such adjustments. It is important that everyone using former ser modules to report the issues that are found. Many thanks to all those involved in integration efforts as well as in new developments that kept the project flying! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

GROUP - change database table
by Daniel W. Graham 28 Dec '12

28 Dec '12

I am attempting to move the location of the GROUP 'grp' table. I created column grp in subscriber table and configured column with same settings as default table and updated column name as follows: modparam("group", "db_url", DBURL) modparam("group", "table", "subscriber") modparam("group", "use_domain", 1) As the other required column names are present in the subscriber table and also default params for this module, I didn't specify them, but I have also tried doing that with same result. I have debugging enabled and when attempting to restart / start kamailio it simply doesn't start and no debug errors are detected, also nothing in log file. If I change to modparam("group", "table", "grp"), kamailio starts fine. Any ideas out there as to what would cause this? Thanks for your time- -dan [cid:image001.gif@01CDDEC4.BE0AD670]

2 4

Is somebody at 29C3 in Hamburg and wants to meet for a drink and some VoIP gossip?
by Klaus Darilion 28 Dec '12

28 Dec '12

regards Klaus

1 0

No Conect
by Matheus Perillo - Rede Itatiaia 27 Dec '12

27 Dec '12

Good afternoon , My name is Matthew and Perillo on IPtel have an account with the username of djperillo . I am not able to connect with anyone. We have to work in the company of various equipment and all with Comrex IPtel account and you can not connect with any of them . Everything indicates to the page Comrex was made and does not connect . The Access are with the SIP Status On LINE . On my iphone the phone symbol on Media5 is green but when I try to connect with the Comrex Access the following message appears " user / password denied , please check your settings and try again . " What can I do to solve this problem ? Thank you . Matheus Perillo Técnico de Manutenção Tel +55 31 2105.3545 Fax +55 31 2105.3677 matheusperillo(a)itatiaia.com.br www.itatiaia.com.br Rua Itatiaia 117 • Bonfim • Belo Horizonte • MG • Brasil • 31210-170

1 0

retransmitted INVITE's
by schwatzi＠web.de 27 Dec '12

27 Dec '12

2 1

extension of SNMPStats module
by 马天 27 Dec '12

27 Dec '12

Hi all, Merry X'mas. I'm working on a project which uses kamailio as sip server. Since I'm really new to kamailio and sip protocool and c, I need some help.My question is that is it possible to extend snmpstats module so the snmpstats module can respond to customized oid.I already wrote the mib file and generated the corresponding c files which handle the request. But I couldn't compile it into snmpstats module.Always get a "No Such Object available on this agent at this OID" when i send GET request using snmpget.Thanks. Regards, Sebastian

2 1

OPTIONS packet from dispatcher module
by Morten Isaksen 27 Dec '12

27 Dec '12

Hi, I have an issue with the OPTIONS packets from the dispatcher module does not contain a max-forward header. This causes the gateway in the other end to not reply to the OPTIONS packet. Can anyone please give a hint where en the code I need to change that. I noticed that dispatcher module uses the tm module to send the OPTIONS but after that I got lost in the code. -- Morten Isaksen

2 1

Problem with RLS
by Dmytro Bogovych 26 Dec '12

26 Dec '12

Hi. I still try to setup RLS capable kamailio instance. Now the server accepts my services and resource list documents. It generated backend subscriptions - i can see it in pua table. When another peer publish its presence - presence tries to send notify to sip address specified in modparam("rls", "server_address", " sip:rls@sip.voipobjects.com"). But the notify does not come back to rls - it is rejected by somewhere in core or other module with error 404. I tried to create the rls(a)sip.voipobjects.com user - it did not help. May anyone give me advice?

1 1

Presence rules?
by Dmytro Bogovych 25 Dec '12

25 Dec '12

Greetings. I try to add rls/xcap presence into resiprocate-based user agent. May anyone tell me what module is responsible for parsing presence rules documents? I'm trying to setup "allow all" presence rules document but did not get success... Thank you!

2 4

Merry Christmas!!!
by Raj Roy Ghandhi 25 Dec '12

25 Dec '12

I WISH Merry Christmas!!! to my Kamailio FAMILY !!!! Best Regards, Roy.

1 0

obsoleting modules_s/*radius modules
by Daniel-Constantin Mierla 24 Dec '12

24 Dec '12

Hello, according to the archive, modules_s/*radius modules were kept because of duplicate and incompatible auth modules at the time there was an attempt to remove them: - http://lists.sip-router.org/pipermail/sr-users/2010-June/064318.html Now there is a single auth modules, thus I guess the modules_s/*radius can be moved to obsolete. If anyone has different opinion, reply with the arguments here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

6 15

Queries
by Ankit sharma 24 Dec '12

24 Dec '12

Hi, I have newly installed Siremis sip router. But I have no idea How I could start or How do I start calls? If you can provide me any Manual so it would be great. Sincerely Ankit

4 3

How to set variable externally
by Mino Haluz 22 Dec '12

22 Dec '12

Hi, I would like to set my custom different debug levels (with flag?) externally with kamctl command. So I neednt restart kamailio if I want to enable/disable debug. Which module should I use in that case? Thanks, Mino

6 12

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