- sr-users - kamailio.org

Re: [Kamailio-Users] Register Request Forward
by toqeer ali 13 Nov '09

13 Nov '09

My rtpproxy works now basically i am using kamailio 1.2 with LCR, but when i dial one number via LCR it dial this number and it works but when i dial second number it says "*filtered destination*", i have to restart the service and then it dial second number... i think LCR not de-alocating the resources any idea ? 2009/11/12 Denis Putyato <denis7979(a)mail.ru> > 1) I would done not ” modparam("nathelper", "rtpproxy_sock", "udp: > 127.0.0.1:8899")” but “modparam("nathelper", "rtpproxy_sock", > "unix:/var/run/rtpproxy.sock")” > > 2) http://www.kamailio.org/docs/modules/1.5.x/nathelper.html#id2468157 > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Thursday, November 12, 2009 2:10 PM > > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks Denis, > > > > It is indeed the problem of NAT. We have tried with live ip and its work > fine. However, behind the NAT it is not working. I have installed RTPproxy , > but still it is not working. Here are the configuration that we use in our > cfg file. > > > > modparam("nathelper","received_avp", "$avp(i:42)") > modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:8899") > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 0) > modparam("nathelper", "sipping_bflag", 7) > modparam("nathelper", "sipping_from", "sip:pinger@127.0.0.1<sip%3Apinger(a)127.0.0.1> > ") > > route { > > ## NAT Detection > # > force_rport(); > if (nat_uac_test("19")) { > if (method=="REGISTER") { > fix_nated_register(); > } else { > fix_nated_contact(); > }; > setflag(5); > }; > > > ................. > > > > and rtpproxy is also configured on the same server. > > > > > > > 2009/11/12 Denis Putyato <denis7979(a)mail.ru> > > No two way communication is a problem of NAT in general. If session has > been established then it’s not codec problem. > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Thursday, November 12, 2009 9:25 AM > > > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > i have a problem with codec. when i call to canada from kamailio to PSTN, i > could not listen his/her voice as they listen( no two way communication) > > but when call to other countries it works... Basically call to canada > support g729 can i define any codec information in kamailio to G729. > > > > Basically the problem is only with local number when you dial with > extension 1 and when you dial with 08 it works... 08 is international > gateway of canada... > > > > X-lite i am using as a softphone. > > > > please help > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > “can i use regular expression in $fu=XXX” > > > > You can use regexp. $fu=~”expression” > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 3:08 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Hmmm > > > > it works now i just remove Record Route function and it works,,, now i can > dial from 602 to 601... thanks :). > > > > i will start work on how i will register multiple extension via kamailio to > asterisk ... can i use regular expression in $fu=XXX instead of actual > extension extension... > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > Registration is a process for telling SoftSwitch (in your variant - > Asterisk) where (on which username, IP address and port) the SoftSwitch must > send INVITE when received incoming call for 601. Try to understand where > your asterisk try to send INVITE when you try to call from 602 to 601. > > ngrep utility will help you (http://ngrep.sourceforge.net/download.html) > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 2:18 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Here is the update: > > > > what i have done so for is that I have "Register extension 601" to asterisk > via kamailio and it is registered!. > > > > When i call to 602 (configured on 192.168.0.1 -- asterisk box) it calls and > make connection and voice call also works (RTP Traffic is ok). > > > > However, when i call to 601 from 602 (which is directly connected to > asterisk without kamailio ), 601 can't receive call via kamailio. This is > the main problem. Please guide me that how i will define this in kamailio > (ip = 192.168.0.2) that xlite with extension 601 can receive call through > 602 which is another extension on Asterisk. > > > > In a nutshell: 601 (registered through Kamailio to Asterisk) ---can call > ---> 602 (registered via Asterisk) > > However: 602----cannot call ------> 601 > > > > please help. > > > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > Try to use dlg_bridge(); function in dialog module > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 12:13 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > > "that client send INVITE to asterisk via kamailio but no replay is in on > this INVITE" > > > > basically Asterisk receive the request and send back to kamailio but > kamailio can't send back to xlite.. > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > «but call does not route back to me» > > > > You mean, that incoming call from asterisk to client via kamailio doesn’t > work or that client send INVITE to asterisk via kamailio but no replay is in > on this INVITE? > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 10:16 AM > > > *To:* Denis Putyato > *Cc:* users@l > > ists.kamailio.org > > > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks Again for the prompt reply... Basicaly i want to route Register > Requests for multiple asterisk via kamailio ... by using pattern matching > of extensions... > > > > is i am on the right path ?? or i can achieve this from dispatcher or load > balancer module .... Basically i want to register specific extensions > to specific asterisk boxes.... > > > > > > from your previous reply i can send call to asterisk and register to as > well but call does not route back to me... any clue ... > > 2009/11/11 Denis Putyato <denis7979(a)mail.r > > “The secret is that the other sip server was asterisk and i want route > call to asterisk via kamailio :).” > > I don’t understand why do you want to do any session with asterisk via > kamailio… not directly to asterisk… but any way the decision is same – > send() or rewritehost() > > > > route { > > if ($fU==”…”&&is_method(“INVITE”)) > > record_route(); > > rewritehost(“192.168.0..1”); > > t_relay(); > > } > > > > Do this before authentication checking is made in kamailio. > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 6:49 AM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks it Registered but i can't call on this .. "authentication Required > from proxy". what i should do now. > > > > The secret is that the other sip server was asterisk and i want route call > to asterisk via kamailio :). > > 2009/11/10 Denis Putyato <denis7979(a)mail.ru> > > Hi > > If not secret, why do you want to use such scheme? > > > > Your decision > > send(); - core function. > > > > { > > If (is_method(“REGISTER”)&&$fU==”….”) send(“udp:192.168.0.1:5060”); > > } > > > > or > > > > { > > If (is_method(“REGISTER”)&&$fU==”….”) { > > rewritehost(“192.168..0.1”); > > t_relay(); > > } > > } > > > > *From:* users-bounces(a)lists.kamailio.org [mailto: > users-bounces(a)lists.kamailio.org] *On Behalf Of *toqeer ali > *Sent:* Tuesday, November 10, 2009 1:56 PM > *To:* users(a)lists.kamailio.org > *Subject:* [Kamailio-Users] Register Request Forward > > > > Hi all, > > > > My question is how to forward Register Request to another sip server. > > > > For example one user (X-lite) want to register with 192.168.0.1(sip > server1) via 192.168.0.2(sip server 2). > > it means that in the domain part of xlite is 192.168...0.2 and > it basically register with 192.168.0.1. > > 192.168.0.2 will forward Register request to 192.168...0.1. > > it will be very helpful if any one can answer. > > > > Thanks > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

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[Kamailio-Users] SKYPE or voipstunt
by Masoom Alam 13 Nov '09

13 Nov '09

whether skype or voipstunt are using SIP protocols. I have experienced that in many countries, where voice ports are blocked by the Telecom providers but still voipstunt and skype works for calling to PSTN. Please give me some links, for further reading on this.

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[Kamailio-Users] Register Request Forward
by toqeer ali 12 Nov '09

12 Nov '09

Hi all, My question is how to forward Register Request to another sip server. For example one user (X-lite) want to register with 192.168.0.1(sip server1) via 192.168.0.2(sip server 2). it means that in the domain part of xlite is 192.168.0.2 and it basically register with 192.168.0.1. 192.168.0.2 will forward Register request to 192.168.0.1. it will be very helpful if any one can answer. Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

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[Kamailio-Users] record-route
by toqeer ali 12 Nov '09

12 Nov '09

Let me describe my scenario: SipUser1 -------register request------------> kamailio (SIP Proxy) ------- forwarded register request-----------> Asterisk 1. In the above scenario, when I register to Asterisk through Kamailio, it works fine when I give the "record-route()" function in the cfg file. Further, using this, I can also call to another extension 602 which is also registered to Asterisk but 602 cannot call to 601 which is registered via kamailio. 2. However, If I remove the "record-route()" function without unregistering 601 and restarting the server, 602 can call to 601 . route { if ($fU=="601"&&is_method("INVITE")) record_route(); rewritehost("10.0.10.111"); //asterisk host route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } should i add anything in script further... -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

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[Kamailio-Users] Listening on more than one port
by Sven Schulz 12 Nov '09

12 Nov '09

Is it possible to have kamailio (1.4.4) to listen on more than one port? Is it as simple as adding another ³port=xxxx² in the config? Sven Schulz Penn State University Telecommunications and Network Services 814.865.6116 sip:sven@psu.edu

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[Kamailio-Users] sending calls to voip provider
by Andres Moya 11 Nov '09

11 Nov '09

Hi, I am with openser about my first 40 hours. So far i managed to install 3.0. Was able to make calls between registered phones. Enabled accounting, etc. Looks i have control ;) Now i have a very simple question. I want to send calls for land lines via voip provider, sipdiscount.com in my case. I want to enable so registered client agent can send invite to that server, but it should re authenticate somehow. I was looking for proxy_authorize www_authorize but it looks like it authenticate my phones only. I was trying rewrite rewritehostport, but of course was getting authentications errors. Is there any manual, previous discussion on this topic? Thank you.

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[Kamailio-Users] change contact address
by GEMINORIUM crux 11 Nov '09

11 Nov '09

hi 1. i want to change contact address .i want to add transport protocol how can i do this ? 2. i am getting Error on $var(query) = "INSERT INTO sip_user(sipuser) VALUES(' "+$fU+"' )"; if(!avp_db_query("$var(query)","$avp(s:test)")){ xlog("ERROR IN Query = $var(query)"); sl_send_reply("400","Bad Request"); exit; } but if i run that query in mysql it give me expected output. why i am getting error ? pls help me ? any suggestion is appreciated. thanks

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[Kamailio-Users] Problem in configuring LCR module
by Masoom Alam 11 Nov '09

11 Nov '09

Hi there, I am a first time user of Kamilio 1.5. We are trying to configure the LCR, we get an error in the syslog. The error in the syslog is: Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:lcr:mod_init: AVP fr_inv_timer_avp has not been defined Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:core:init_mod: failed to initialize module lcr Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:core:main: error while initializing modules We have already configured FR_inv_timer_avp parameter like this in the Kamilio.cfg: modparam("tm", "fr_inv_timer_avp", "$avp(i:70") The "tm" module is already loaded in the Kamilio.cfg. Please guide, where we are making problems?

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[Kamailio-Users] Translate "486 Busy Here" to "600 Busy Everwhere"
by geoffreymina＠gmail.com 11 Nov '09

11 Nov '09

Hello, I have a need to translate a 486 from my asterisk server to a 600 when sending upstream to my carrier. The problem I am seeing is that when I return 486, the carrier re-presents the call to my next server in the SRV rotation... I really want the 486 to be a final disposition, so I am assuming sending a 600 would solve my issue. I am not sure how to go about forwarding the message, but modifying the SIP code... is that even allowed? This would be something like what I'm trying to do: failure_route[1]{ if(t_check_status("

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Re: [Kamailio-Users] Translate "486 Busy Here" to "600 Busy Everwhere"
by Geoffrey Mina 10 Nov '09

10 Nov '09

Apologies... sent directly to Alex and not the list. On Tue, Nov 10, 2009 at 12:02 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: > Alex, > Thanks for the info. We are on the same page about what a proxy > should, and should not do... however, I am strugging with my ITSP > (Level3) a bit on this. Unfortunately, Asterisk is my media end-point > here and there is no mechanism in Asterisk to send back a 600. Level3 > is exhibiting very odd behaivor, and I have a ticket open with them, > but who knows if it will ever go anywhere. > > CallID: 1627795841_10332(a)4.55.4.227 > Level3 --> INVITE --> Kamailio --> Asterisk > Asterisk --> 183 Progress --> Kamailio --> Level3 > Asterisk --> 486 Busy --> Kamailio --> Level3 > Level3 --> ACK --> Kamailio --> Asterisk > > [at this point Level3 IMMEDIATELY sends me another presentation of the > call with a different CallID and the process repeats] > > CallID: 604627041_120490736(a)4.55.20.99 > Level3 --> INVITE --> Kamailio --> Asterisk > Asterisk --> 183 Progress --> Kamailio --> Level3 > Asterisk --> 486 Busy --> Kamailio --> Level3 > Level3 --> ACK --> Kamailio --> Asterisk > > > I am pretty sure I am in the right here, and Level3 shouldn't be > failing over in a 486 scenario... unfortunately my customer doesn't > care why our systems aren't talking properly :) > > Thanks! > > > > On Tue, Nov 10, 2009 at 10:38 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> geoffreymina(a)gmail.com wrote: >>> >>> Hello, >>> I have a need to translate a 486 from my asterisk server to a 600 when >>> sending upstream to my carrier. The problem I am seeing is that when I >>> return 486, the carrier re-presents the call to my next server in the SRV >>> rotation... I really want the 486 to be a final disposition, so I am >>> assuming sending a 600 would solve my issue. >>> >>> I am not sure how to go about forwarding the message, but modifying the >>> SIP code... is that even allowed? >>> >>> This would be something like what I'm trying to do: >>> >>> failure_route[1]{ >>> if(t_check_status(" >> >> Sure, you can do that. It is not a good idea to do so in principle because >> it violates how proxies should behave; proxies are very thin interoperation >> layers that are designed to statefully pass requests and replies back and >> forth more or less strictly as intended by the endpoints involved, and a >> B2BUA would more ideally fit your goal. Nevertheless, it is possible. >> >> failure_route[1] { >> >> if(t_was_cancelled()) >> exit; >> >> if(t_check_status("486")) { >> t_reply("603", "Declined"); >> exit; >> } >> } >> >> >> -- >> Alex Balashov - Principal >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> >

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[Kamailio-Users] Double acc records in DB
by Alex Hermann 10 Nov '09

10 Nov '09

Hello, I noticed that sometimes I get 2 identical acc records in the database, although I get only 1 in the syslog (I use both db_flag and log_flag). Of course, the second record in the DB shouldn't be there. There are no retransmissions of neither request nor response and only a single branch exists. How is this possible, and more importantly, how can I prevent the double records? This is on Kamailio 1.4+svn. -- Greetings, Alex Hermann

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[Kamailio-Users] Question about translating 486 to 600
by Geoffrey Mina 10 Nov '09

10 Nov '09

Hello, I have a need to translate a 486 from my asterisk server into a 600 to my ITSP. I haven't figured out a way to force asterisk to send the 600, so I guess I need to rely on Kamailio to do that for me. The problem is that when the INVITE is sent to me, if I return a 486, my ITSP immediately sends a new INVITE to the next gateway in my SRV records. I want the 486 to be a final disposition in this scenario, so I am guessing I need to send 600 for the carrier to not try another gateway. How would I go about doing this? failure_route[1]{ if(t_check_status("486")){ t_relay("600","Busy Here"); } } I know the above example isn't valid... but that is basically what I need to accomplish. Thanks!

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[Kamailio-Users] avp
by Anders 10 Nov '09

10 Nov '09

Hi all, Here I am - back with more "Kamailio 101"-questions! ;-) I need to strip the first character from the uri and write this to the 'acc' table. What I'm doing so far in kamailio.cfg is: modparam("acc", "db_extra", "............., $avp(s:user_ip_tag)"); So I've added the avp-thing in that line. And independently I've created a new column in the 'acc' table called "user_ip_tag". If the above is correct, my next step is to write something to this variable. In the routing, I have this: if (uri=~"^sip:[1-9]00*") { do_something_here_to_take_first_character_of_the_uri_and_put_to_avp...but_how? strip(1); } What I want to do, is to take the 1-9 digit, if it starts with that and has to zeroes after, and put that to the avp - and the strip it off before I move on in the routing. Yes, I have browsed the documentation, but I'm not sharp enough to understand it, it seems. (or alternatively, the documentation is not made for beginners). Thanks!

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[Kamailio-Users] accounting
by Rajesh 09 Nov '09

09 Nov '09

Dear all, I would like to know if i can get the duration between two calls in Accounting module..I am in an attempt to do "Billing" module for OpenSER and hence would like to know the same. -- Regards, Rajesh

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SR or Kamailio.
by Vicente 09 Nov '09

09 Nov '09

Hi, This is my first message to the list. So, sorry if I say something stupid. I've installed sip-router from GIT repository. It is now compiled and installed. I can see that bins ser, sercmd and ser_mysql.sh has been copied to /usr/local/sbin. ser runs properly but sercmd tries to use /tmp/ser_ctl (ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory [2]) while /temp/sip-router_ctl is the socket generated by ser. ser_mysql.sh also fails with error: ./ser_mysql.sh: 326: Bad substitution The INSTALL file on the repository instructs to use serctl utility to watch server health but no binary has been generated with that name. Is it replaced by sercmd? What is the status of sip-router.org? Is it ready for production? Or is it a better option to use Kamailio at the moment? Best regards.

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[Kamailio-Users] How to recognize the caller ID
by Masoom Alam 09 Nov '09

09 Nov '09

Hi Every one, Is the following scenario possible and if yes, how? I want to identify the caller ID inside the Kamailio so that a specific caller can be routed to an Asterisk box. Actually, the caller -- a SIP client or an IP phone client is not registered at Kamailio, but Kamailio has to act as a Proxy to a group of Asterisk servers. The sketch in my mind is to use some if-then-else statements within the [route] where user can call a particular number say NXXXXXXX201 and then the Kamailio can strip off the first number and re-route it to an Asterisk server. Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

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[Serusers] 500 internal error
by Sabatella, Michael 08 Nov '09

08 Nov '09

Hi, I am getting a 500 internal server error. I am not sure why as my config did not change. I do not want to restart the ser proxy. Could I get a list of causes for a 500 server error? Thanks, Michael Sabatella Network Services Engineer Office: 1-201-253-2280 Cell: 1-908-380-5321 ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

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[Kamailio-Users] A architectural question
by Masoom Alam 08 Nov '09

08 Nov '09

Hi there, Is it possible that Kamailio is used in front of a group of Asterisk servers. That is, Kamailio as a REGISTRAR Translation can proxy the SIP registration and identify which PBX to send on LAN by the ID before the EXT underlined from IP PHONE. Please advise with some links through which it is possible to handle such an architecture. Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

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[Kamailio-Users] OCS?
by Jay R. Worthington 06 Nov '09

06 Nov '09

Hi, has someone managed to "federate" kamailio with an Office Communications Server? I'm mainly looking for IM&Presence (to gate into ejabberd), voice would be nice to have... Regards, Jay

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[Kamailio-Users] routing question
by Andrew Gallo 06 Nov '09

06 Nov '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings: Hope you don't mind a question from someone just getting his feet wet :) First off, if I want Kamailio to act as a SIP router among, say, a group of Asterisk PBXs (meaning, Kamailio will have no user registrations, but just pass SIP messages to and from the Asterisks), would I do this in the least cost routing table or in carrier routes table, or via another method? (assume for the moment that all the PBXs are under my control, these aren't connections outside my organization, no connections to the PSTN). I figured it would be with lcr, so I tried compiling 3.0.0RC1 with that module removed from the exclude list. I received the following error on 'make install' make[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' config.mak included make[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 - -minline-all-stringops -falign-loops -ftree-vectorize - -fno-strict-overflow -mtune=athlon64 -Wall -DNAME='"kamailio"' - -DVERSION='"3.0.0-rc1"' -DARCH='"i386"' -DOS='linux_' - -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux - -DSER_VER=3000000 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC - -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP - -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER - -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS - -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 - -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN - -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL - -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL - -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT - -I/usr/local/include -I/usr/local/include -I/opt/include - -I/usr/sfw/include -DOPENSER_MOD_INTERFACE -DMOD_NAME='"lcr"' -c hash.c - -o hash.o In file included from hash.c:26: lcr_mod.h:34:18: error: pcre.h: No such file or directory In file included from hash.c:26: lcr_mod.h:51: error: expected specifier-qualifier-list before âpcreâ hash.c:34: error: expected declaration specifiers or â...â before âpcreâ hash.c: In function âlcr_hash_table_insertâ: hash.c:56: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:56: error: âfrom_uri_reâ undeclared (first use in this function) hash.c:56: error: (Each undeclared identifier is reported only once hash.c:56: error: for each function it appears in.) hash.c:58: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:59: error: âstruct lcr_infoâ has no member named âfirst_gwâ hash.c:60: error: âstruct lcr_infoâ has no member named âpriorityâ hash.c:66: error: âstruct lcr_infoâ has no member named ânextâ hash.c: In function âlcr_hash_table_contents_freeâ: hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:107: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:108: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:110: error: âstruct lcr_infoâ has no member named ânextâ make[1]: *** [hash.o] Error 1 make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' make: *** [install-modules] Error 1 If I leave lcr in the exclude list, I can make/make install, but if I try to add an entry into the lcr table, I get this error: # kamctl lcr addgw labCM 172.16.59.6 5060 1 2 1 ERROR 1054 (42S22) at line 1: Unknown column 'ping' in 'field list' ERROR: lcr - SQL Error System info: Kamailio 3.0.0 RC1 from GIT debian 5.0.3 (2.6.26-2-686) gcc -v Using built-in specs. Target: i486-linux-gnu Configured with: ../src/configure -v --with-pkgversion='Debian 4.3.2-1.1' --with-bugurl=file:///usr/share/doc/gcc-4.3/README.Bugs - --enable-languages=c,c++,fortran,objc,obj-c++ --prefix=/usr - --enable-shared --with-system-zlib --libexecdir=/usr/lib - --without-included-gettext --enable-threads=posix --enable-nls - --with-gxx-include-dir=/usr/include/c++/4.3 --program-suffix=-4.3 - --enable-clocale=gnu --enable-libstdcxx-debug --enable-objc-gc - --enable-mpfr --enable-targets=all --enable-cld - --enable-checking=release --build=i486-linux-gnu --host=i486-linux-gnu - --target=i486-linux-gnu Thread model: posix gcc version 4.3.2 (Debian 4.3.2-1.1) Thank you! -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkrx/CMACgkQQr/gMVyFYySWQgCeJoW2VLWRI85bHdC9ThwnEwLc pWQAoJZVtBEzdRI8+3xBoLnlWJxNXbj/ =Wwjk -----END PGP SIGNATURE-----

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[Kamailio-Users] Dispatcher module question
by DJB 05 Nov '09

05 Nov '09

Is there a way that I can pass variable into ds_select_dst(set, alg) on the set parameter so I can query from database table for setid; for instance, ds_select_dst("avp(customer_setid)", "4"). I've tried, but it did not work. Please advise. Thank you.

3 5

difference between memlog and memdbg
by Klaus Darilion 05 Nov '09

05 Nov '09

Hi! What's the difference betwenn memlog and memdbg? thanks klaus

2 2

[Kamailio-Users] uac_replace_from check
by jh＠tutpro.com 05 Nov '09

05 Nov '09

if i have understood correctly, if uac_replace_from() is called in "auto" mode, encoded from uri is added into an extra param of r-r header when the request is forwarded towards uas. does the module check that reply to the request from uas really includes the added parameter in its r-r header or is correct operation of in-dialog requests at the mercy of the uas? -- juha

7 37

Re: [Kamailio-Users] LCR problem with "id" column and match priorities
by jh＠tutpro.com 05 Nov '09

05 Nov '09

it turned out that the problems ricardo reported on gw selection in lcr module were due to the fact that he has the same gw (same in terms of ip address) in more than one gw group and in each group the gw has different trip/tag attributes. load_gws() function tries the same gateway only once, but it is not currently well defined from which group gw's attributes are taken when request is sent to the gateway. currently the attributes are taken from the gw group, which happens to be the first in in-memory gw table. one could think that this is a feature, i.e., if the same gw is in many gw groups, it must have the same attributes in every group. the other alternative is that this is a bug, i.e., same gw's attributes can differ from group to group and the ones that are used must be the ones of the chosen group. i have a fix written for sr_3.0 lcr module, but i'm hesitating to introduce the fix also to version 1.5, because the fix has not been tested well yet. any opinions on what should be done? -- juha

3 9

Re: [Kamailio-Users] syntax error
by Daniel-Constantin Mierla 05 Nov '09

05 Nov '09

don't forget -- always cc the mailing list. On 05.11.2009 5:39 Uhr, GEMINORIUM crux wrote: > Daniel, > i am transferring tls protocol to udp protocol, > sorry, i don't mention it. > > client <---tls----> kamilio <----udp---> asterisk > > thanks for your reply. > > > On Wed, Nov 4, 2009 at 7:38 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 04.11.2009 14:34 Uhr, GEMINORIUM crux wrote: > > hi, > i am new to kamailio, > > getting syntax error: > $var(test) = "udp:"+$rd+":"+$rp; > t_relay_to_udp($var(test)); > > t_relay_to_udp($rd,$rp); > > working with string. > > is there any reason ? how to resolve it ? > i have to use database so i need to use variable. > > any suggestion is appreciated. > > you have to do: > > $du = "sip:"+$rd+":"+$rp; > t_relay(); > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > * http://www.asipto.com/ > > -- Daniel-Constantin Mierla * http://www.asipto.com/

2 3

[Kamailio-Users] vim syntax file for 3.0
by Daniel-Constantin Mierla 04 Nov '09

04 Nov '09

Hello, I updated the vim syntax file for Kamailio 3.0 and SR3.0/devel. I attached as well a script to auto-detect the type of .cfg file based on matching compatibility string (i.e., #!XYZ), loadmodule or main route line. To install, copy the files in your home dir - see the hints inside each file to know which one to place where: .vim/syntax/ser.vim .vim/ftdetect/ser.vim If you don't install the file type detect script, then you can use ":setf ser" once you opened the config file. Enjoy, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ " Copy this file to $HOME/.vim/ftdetect/ser.vim func! s:cfgType() let max = line("$") > 200 ? 200 : line("$") for n in range(1, max) if getline(n) =~ '^\s*#!$KAMAILIO\|OPENSER\|SER\|ALL\|MAXCOMPAT$' set filetype=ser return elseif getline(n) =~ '^\s*modparam\s*(\s*"[^"]\+"' set filetype=ser return elseif getline(n) =~ '^\s*route\s*{\s*' set filetype=ser return endif endfor setf cfg endfunc au BufNewFile,BufRead *.cfg call s:cfgType() " -*- vim -*- " FILE: ser.vim " LAST MODIFICATION: 2009-05-28 18:30 " (C) Copyright 2008 Stanisław Pitucha <viraptor(a)gmail.com> " (C) Copyright 2009 Daniel-Constantin Mierla <miconda(a)gmail.com> " Version: 1.01 " USAGE: " " Save this file to $VIMFILES/syntax/ser.vim. Either add a detection " script to filetypes.vim, or set filetype manually to "ser" when " editing SIP Router configuration file with 'setf ser'. " " List of keyword and core functions taken from latest dev version of " SIP Router. Module functions not included. " " Tested only on vim 7.1 " " Example: "setf ser" " " REQUIREMENTS: " vim (>= 7) if exists("b:current_syntax") finish endif syn match serConfigParamLine '^[^=]\+=.*$' contains=serCoreParameter,serString,serConfigConstant,serSpecial,serNumber,serCppComment,serHashComment syn region serConfigModparam start='^\s*modparam\s*(' end=')' contains=serString,serNumber syn match serConfigModule '^\s*loadmodule\s*"[^"]\+"' contains=serString syn keyword serTodo TODO FIXME XXX contained syn match serOperator '!\|&&\|||\|=[~=]\?\|>\|<\|+\|-\|/\|\*\||\|&\|^\|\~\|defined\|eq\|ieq\|ne\|ine\|mod' display contained syn region serCppComment start='/\*' end='\*/' contains=serTodo syn match serHashDefine '#!define\s\|#!ifdef\s\|#!ifndef\s\|#!endif\|#!else\|#!KAMAILIO\|#!OPENSER\|#!SER\|#!MAXCOMPAT\|#!ALL' " syn match serHashDefine '^\s*#!.+$' syn match serHashComment '#[^!].*$\|#$' contains=serTodo syn match serStringEscape '\\.' contained syn match serNumber '[0-9]\+' contained syn region serString matchgroup=Normal start='"' skip='\\"' end='"' contained contains=serVariable,serStringEscape syn match serVariable "$[a-zA-Z_][a-zA-Z0-9_]*$([^)]\+)$\?" contained syn match serIdentifier '[a-zA-Z_][a-zA-Z0-9_]*' contained syn keyword serStatement route if else switch case default break exit return drop while include_file contained syn keyword serSpecial yes no on off true false enabled disabled contained syn keyword serCoreKeyword af dst_ip dst_port from_uri method msg:len proto status snd_af snd_ip snd_port snd_proto src_ip src_port to_af to_ip to_port to_proto to_uri uri uri:host uri:port contained syn keyword serCoreValue udp UDP tcp TCP tls TLS sctp SCTP inet INET inet6 INET6 sslv23 SSLv23 SSLV23 sslv2 SSLv2 SSLV2 sslv3 SSLv3 SSLV3 tlsv1 TLSv1 TLSV1 max_len myself contained syn keyword serCoreFunction forward forward_tcp forward_udp forward_tls forward_sctp send send_tcp log error exec force_rport add_rport force_tcp_alias add_tcp_alias udp_mtu udp_mtu_try_proto setflag resetflag isflagset flags bool setavpflag resetavpflag isavpflagset avpflags rewritehost sethost seth rewritehostport sethostport sethp rewritehostporttrans sethostporttrans sethpt rewriteuser setuser setu rewriteuserpass setuserpass setup rewriteport setport setp rewriteuri seturi revert_uri prefix strip strip_tail userphone append_branch set_advertised_address set_advertised_port force_send_socket contained syn keyword serCoreParameter debug fork log_stderror log_facility listen alias auto_aliases dns rev_dns dns_try_ipv6 dns_try_naptr dns_srv_lb dns_srv_loadbalancing dns_udp_pref dns_udp_preference dns_tcp_pref dns_tcp_preference dns_tls_pref dns_tls_preference dns_sctp_pref dns_sctp_preference dns_retr_time dns_retr_no dns_servers_no dns_use_search_list dns_search_full_match dns_cache_init use_dns_cache use_dns_failover dns_cache_flags dns_cache_negative_ttl dns_cache_min_ttl dns_cache_max_ttl dns_cache_mem dns_cache_gc_interval dns_cache_del_nonexp dns_cache_delete_nonexpired dst_blacklist_init use_dst_blacklist dst_blacklist_mem dst_blacklist_expire dst_blacklist_ttl dst_blacklist_gc_interval port statistics maxbuffer children check_via phone2tel syn_branch memlog mem_log memdbg mem_dbg sip_warning server_signature reply_to_via user uid group gid chroot workdir wdir mhomed disable_tcp tcp_children tcp_accept_aliases tcp_send_timeout tcp_connect_timeout tcp_connection_lifetime tcp_poll_method tcp_max_connections tcp_no_connect tcp_source_ipv4 tcp_source_ipv6 tcp_fd_cache tcp_buf_write tcp_async tcp_conn_wq_max tcp_wq_max tcp_rd_buf_size tcp_wq_blk_size tcp_defer_accept tcp_delayed_ack tcp_syncnt tcp_linger2 tcp_keepalive tcp_keepidle tcp_keepintvl tcp_keepcnt tcp_crlf_ping disable_tls tls_disable enable_tls tls_enable tlslog tls_log tls_port_no tls_method tls_verify tls_require_certificate tls_certificate tls_private_key tls_ca_list tls_handshake_timeout tls_send_timeout disable_sctp enable_sctp sctp_children sctp_socket_rcvbuf sctp_socket_receive_buffer sctp_socket_sndbuf sctp_socket_send_buffer sctp_autoclose sctp_send_ttl sctp_send_retries advertised_address advertised_port disable_core_dump open_files_limit shm_force_alloc mlock_pages real_time rt_prio rt_policy rt_timer1_prio rt_fast_timer_prio rt_ftimer_prio rt_timer1_policy rt_ftimer_policy rt_timer2_prio rt_stimer_prio rt_timer2_policy rt_stimer_policy mcast_loopback mcast_ttl tos pmtu_discovery exit_timeout ser_kill_timeout max_while_loops stun_refresh_interval stun_allow_stun stun_allow_fp server_id description descr desc loadpath mpath contained syn region serBlock start='{' end='}' contained contains=serBlock,@serCodeElements syn match serRouteBlock '$failure_\|onreply_\|branch_\|event_\|onsend_$\?route$\s*\[[^\]]\+\]$\?' contained contains=serNumber,serString,serIdentifier syn region serRrouteBlockFold matchgroup=serRouteBlock start="$failure_\|onreply_\|branch_\|event_\|onsend_$\?route$\s*\[[^\]]\+\]$\?\s*\n\?{" matchgroup=NONE end="}" contains=serBlock,@serCodeElements syn cluster serCodeElements contains=serHashDefine,serCppComment,serHashComment,serNumber,serString,serVariable,serOperator,serStatement,serKeyword,serCoreKeyword,serCoreValue,serCoreFunction,serIdentifier hi def link serCppComment Comment hi def link serHashComment Comment hi def link serHashDefine Special hi def link serTodo Todo hi def link serConfigModparam Function hi def link serConfigModule Keyword hi def link serKeyword Keyword hi def link serCoreKeyword Special hi def link serCoreValue Special hi def link serCoreFunction Function hi def link serRouteBlock Type hi def link serRrouteBlockFold Type hi def link serIdentifier Identifier hi def link serSpecial Special hi def link serCoreParameter Keyword hi def link serOperator Operator hi def link serStatement Conditional hi def link serNumber Number hi def link serVariable Identifier hi def link serString String hi def link serStringEscape Special let b:current_syntax = "ser"

2 2

[Kamailio-Users] Social Networking Event
by Daniel-Constantin Mierla 04 Nov '09

04 Nov '09

Hello, several folks working with Kamailio, SIP Router, SER, OpenIMSCore, SEMS and Asterisk are in Berlin next week, so we think of having a dinner (or beer) meeting Thursday, 19:00, Nov 12, 2009. If happens that you are around and want to join, please send me an email to make sure you get a seat. As usual for this kind of event, free participation with everyone paying for himself/herself. Just to have an idea about how it could be, see some photos from previous similar events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ From private discussions so far, we should be already about 10 people, confirmed including myself, Olle E. Johansson (Asterisk) and Elena-Ramona Modroiu. There is a lot to celebrate, from integration work within SIP Router Project to upcoming Kamailio 3.0 (now RC1). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] [Fwd: Re: uac_replace_from again]
by marius zbihlei 04 Nov '09

04 Nov '09

Forgot to forward to list

1 0

[Kamailio-Users] syntax error
by GEMINORIUM crux 04 Nov '09

04 Nov '09

hi, i am new to kamailio, getting syntax error: $var(test) = "udp:"+$rd+":"+$rp; t_relay_to_udp($var(test)); t_relay_to_udp($rd,$rp); working with string. is there any reason ? how to resolve it ? i have to use database so i need to use variable. any suggestion is appreciated. thanks

2 1

[Kamailio-Users] Kamailio 1.5.0 and NAT
by Путято Денис 04 Nov '09

04 Nov '09

Hello everybody I have such problem with kamailio and client which behind NAT This is REGISTER message which came from client. This message came from public IP (for example, 1.1.1.1) and source port 44197. (This information I got from sip_trace) REGISTER sip:sip.ptl.ru:5060;transport=UDP SIP/2.0 From: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=a82eda3-13c4-47798948-1a1296-5259a46 To: <sip:xxxxxxx@sip.ptl.ru:5060> Call-ID: a82eda3-13c4-47798948-1a1296-9641c88(a)sip.ptl.ru CSeq: 446229198 REGISTER Via: SIP/2.0/UDP 10.130.237.163:5060;branch=z9hG4bK-47799e27-6b9841-84448db Max-Forwards: 70 Supported: timer Contact: <sip:xxxxxxx@10.130.237.163:5060;transport=UDP> Expires: 60 Content-Length: 0 This is Unauthorized message was sent by kamailio to 1.1.1.1 and: destination port not 44197 (as I expected) but 5060!!! SIP/2.0 401 Unauthorized From: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=a82eda3-13c4-47798948-1a1296-5259a46 To: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=fdc454a3cdd8f9e5718382b74f33aeed.3b81 Call-ID: a82eda3-13c4-47798948-1a1296-9641c88(a)sip.ptl.ru CSeq: 446229198 REGISTER Via: SIP/2.0/UDP 10.130.237.163:5060;branch=z9hG4bK-47799e27-6b9841-84448db;received=1.1.1.1 WWW-Authenticate: Digest realm="domain name", nonce="4af0129b000000047c1020258f9f0efc4de1b76663dd2ab4" Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 And as you understand 401 reply, which was sent to 1.1.1.1:5060 can't reach client's equipment. A piece of kamailio.cfg relating to REGISTER processing: route(0) { :. if (!is_method("INVITE")) route(8); :. if (is_method("REGISTER")) { route(11); exit; } :.. } route(8) { if (nat_uac_test("5")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } route(11) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if (isflagset(5)) setbflag(6); if (!save("location")) { sl_reply_error(); exit; } } Hearty thanks for any help.

2 1

[Kamailio-Users] 404 sparks next gw and 503
by Anders 03 Nov '09

03 Nov '09

Hi, This list becomes a mix of high-level tech and low-level beginner questions... - and I'm in the latter group. Making two lists might be over-kill, though. Here's my issue: My call termination provider is rightfully returning a 404 for a call to a number that doesn't exist. This currently seems to spark Kamailio to look for the next gw, but in this case there is no next gw set up in the lcr/gw - so it returns a 503 to the caller. If I'm confident that my provider is right that there is no such number, and 404 is the correct answer, wouldn't it be better to return a 404 to the caller rather than a 503? With 503 stats also get screwed up, because my client can just see a bunch of 503s, which is not a very detailed picture of the real situation. If I should return a 404, how would I go about this? What would I do in cfg to ensure that a 404 return does not spark it to look for a next gw, but sends 404 back to the caller? Alternatively, I could try to figure out how to get the return from my provider saved to the acc table on the INVITE record, and I'd have something more detailed to run stats on. Does that make sense? Thanks!

3 3

[Kamailio-Users] OpenSER/Kamailio behind NAT
by systeo 02 Nov '09

02 Nov '09

Hi, I successfully installed Kamailio on the Amazon computing cloud (EWS), and it runs fine. However, I can't get to it (register) from any UACs from the public internet. In amazon, all the instances (i.e. "virtual machines) are in a "private" IP, kindda like the private IP you get in a LAN. However, Amazoin gives you a "public IP" that can be mapped to any machine instance in your account, i.e. the machine that kamailio is running. However, when i use this public IP in the configuration of the UACs, it cannot get to it. How can I get kamailio to listen on this public IP? Secondly, and related to the first, when I run kamailio in my LAN (behind a router), say on 192.168.1.113, and all my UACs are in the same LAN, say 192.168.1.112, 192.168.1.114, etc, I can register ok and make calls just fine. However, when I try to come in from the public internet, no UACs can register to my Kamailio server (running on 192.168.1.113), even though I am forwarding port 5060 from my router to 192.168.1.113 (where kamailio is running). I think this problem is related to the first one, in that I cannot get Openser (kamailio) to listen on a public IP, when it is behing a "LAN". Can anyone help? Lastly, the same issue with rtpproxy also behind the LAN, but it needs a pulic IP, right? Please help...anyonw. Hassen BEN SLIMA Analyst developer TEL : 00216 74 863 507 FAX : 00216 74 863 506 E-MAIL : <mailto:tarak@systeo.biz> hassen(a)systeo.biz Web : <http://www.systeo.biz> www.systeo.biz Adresse : Route de Tunis km 10, Technopôle de Sfax - 3021 - Sakiet EZZIT - Sfax - Tunisie <http://www.systeo.biz/> Sans titre-4

2 1

[Kamailio-Users] force_rtp_proxy() usage with IP that is not natively homed
by Alex Balashov 02 Nov '09

02 Nov '09

Greetings, Please excuse the apparent silliness of this request, but I am curious if there is a combination of flags passed to force_rtp_proxy() and/or rtpproxy invocation options that can allow force_rtp_proxy() to substitute an SDP endpoint IP that is not a physical device/interface on the host on which rtpproxy is running? Let me explain further: 1. I have a host with a single network interface that has private IP 192.168.1.254. 2. I am receiving an inbound call from outside the network, coming in on a public IP address 4.4.4.4 that is a 1-to-1 DNAT/SNAT to 192.168.1.254. 3. The call is forwarded to a PBX on the same LAN as the native interface, e.g. 192.168.1.250. 4. I would like to set up SDP in such a way that 192.168.1.250 sendsmedia to the 192.168.1.254 interface while the outside host initiating the call sends its media to 4.4.4.4. Under normal circumstances in which both the public and private interface were physically present on the machine, this would call for the use of "bridging" mode (rtpproxy invoked with -l 4.4.4.4/192.168.1.254 and use of "ie" and "ei" flags with force_rtp_proxy() - we have previously established that rtpproxy_offer/answer do not actually seem to work as advertised). But is it possible to "bridge" to an IP that is not routed to the box? I expected that force_rtp_proxy()'s second parameter - ip_address - would do this. However, this does not appear to let me override the SDP endpoint with any IP address I want; it seems that I am still limited to choosing from those IPs physically available on the rtpproxy host. I would guess that this is because ip_address is passed as a parameter through rtpproxy's control API socket and is validated by rtpproxy itself, and if it is not a local existing address, is overridden with one that is. Is there a clever combination of flags that can be used to accomplish this, or is it a lost cause? Thanks! -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

3 6

[Serusers] error: SER/FIFO not accessible: 2
by Brian Artigas 02 Nov '09

02 Nov '09

Hello, I'm trying to add a user to the database and getting the following error back: error: SER/FIFO not accessible: 2 Here is an excerpt from my ser.cfg file: debug=3 fork=no log_stderror=yes listen=192.168.1.30 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo_mode=0777 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) Anyone have any ideas? Thanks, Brian

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[Kamailio-Users] Broken Debian packaging in Kamailio 3.0.0-RC
by Kim Philby 31 Oct '09

31 Oct '09

Hi! I've tried to make Debian packages for latest Kamailio, but some issues arised. First, packages are named ser-stuff, shouldn't they be called kamailio-stuff even though Kamailio now uses SIP-Router core? Also, Debian package making seems to be broken: # install the *.sql files for ser_mysql.sh script cp -f scripts/mysql/*.sql /home/kmphlb/src/kamailio-3.0.0/kamailio/debian/ser/usr/lib/ser/ cp: target «/home/kmphlb/src/kamailio-3.0.0/kamailio/debian/ser/usr/lib/ser/» is not a directory make[1]: *** [install] Error 1 Directories are named 'ser'. Shouldn't it be just 'kamailio'? Regards, -- Kim Philby

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[Kamailio-Users] Kamailio 3.0.0 RC1
by Daniel-Constantin Mierla 30 Oct '09

30 Oct '09

Hello, I created a tarball with sources of Kamailio 3.0.0 RC1 in order to easy the usage and testing: http://www.kamailio.org/pub/kamailio/3.0.0-rc1/src/kamailio-3.0.0-rc1_src.t… There is also now a tutorial to guide the installation of kamailio 3.0 branch from GIT repository: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git Two wiki pages were created to collect what is new in Kamailio 3.0 and how to migrate to this version: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 Both documents need a lot of care since many new features are missing or are not detailed, and definitely update guidelines do not cover all changes. Help is very much appreciated there. As we keep hunting and fixing issues, improve documentation, I hope to get to full release Kamailio 3.0.0 very soon. Enjoy RC1 meanwhile and report any issue you find to sr-dev(a)lists.sip-router.org. Thanks, Daniel

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[Kamailio-Users] Route header problem
by Alex Rendour 30 Oct '09

30 Oct '09

Hello, We have a problem with route on header. I receive an ack from X.X.X.X (asterisk) on Y.Y.Y.Y (kamailio 1.5.3). Y.Y.Y.Y relay ACK to Z.Z.Z.Z Y.Y.Y.Y this route from X.X.X.X: <sip:Y.Y.Y.Y;lr=on;ftag=as7e58db45>,<sip:11dc0d7918b56374@Z.Z.Z.Z;lr> And forward as the same to Z.Z.Z.Z. The route header must be : <sip:11dc0d7918b56374@Z.Z.Z.Z;lr> I think my kamailio must delete the first part of the route header, like this explaination: http://www.tech-invite.com/Ti-sip-dialog.html#inv-ack Could you help me? Regards, -- alex

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[Kamailio-Users] kam 1.5 syslog on solaris 10
by Robert McGilvray 29 Oct '09

29 Oct '09

I'm having some issues getting kam to log correctly to syslog after it daemonizes. I get all the initialization entries in the proper logfile but nothing after that - I have xlog statements throughout the config which only show up in syslog if fork=no. I've tested my syslog config with the system logger which works correctly executed as the kamailio user. If I run syslog in debug I don't see it receiving anything from kam after initialization. Any ideas? log_stderror=no log_facility=LOG_LOCAL1 debug=7 fork=yes children=4 /etc/syslog.conf (with tabs) local1.debug /var/log/globeop/kamailio.log This is the correct format of syslog in solaris. It only uses wildcard (*) in facility not levels. Specifying debug will include everything at higher levels. From the man page: For a given facility and level, syslogd matches all messages for that level and all higher levels. For example, an entry that specifies a level of crit also logs messages at the alert and emerg levels. This config is working correctly on sol 10 with openser 1.3. ykt1sipprd2# tail -f /var/log/globeop/kamailio.log Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 951507 local1.debug] DBG:core:grep_aliases: comparing host [2:loghost:5060] with us [1:loghost:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 951507 local1.debug] DBG:core:grep_aliases: comparing host [2:loghost:5060] with us [1:ykt1sipprd2:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 793175 local1.debug] DBG:core:grep_aliases: no match for: [2:loghost:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 634971 local1.debug] DBG:core:socket2str: <tcp:172.20.151.148:5060> Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 797031 local1.debug] DBG:core:fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 596959 local1.debug] DBG:core:fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16560 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 362601 local1.debug] DBG:core:fm_malloc_init: params (fc400000, 33554432), start=fc400000 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 551434 local1.debug] DBG:core:shm_mem_init_mallocs: success Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 123545 local1.info] INFO:core:init_tcp: using /dev/poll as the TCP io watch method (auto detected) Oct 27 09:50:46 ykt1sipprd2 kamailio[3815]: [ID 721503 local1.warning] WARNING:core:daemonize: pid file contains old pid, replacing pid -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] How to take MAC address
by Chandrakant Solanki 29 Oct '09

29 Oct '09

Hi I want to know the MAC address of registered client on kamailio... suppose i have 3 Client registered with 192.168.1.2, 192.168.1.3 and 192.168.1.4 from these 3 Client, i am registered successfully with 101, 102, 103(a)test.com But for additional authentication ... i also need MAC address for this Client. Is there any module, method, function or variable to take MAC address or is there any alternative for the above case. Waiting for favorable reply... -- Regards, Chandrakant Solanki

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[Kamailio-Users] Pass calls from/to another realm to gateway
by Alexander 29 Oct '09

29 Oct '09

I've encountered one problem I can not solve :( The situation is following: we've got Kamailio working together with our own SIP platform. Our platform is about various call processing business logic and billing. All calls from SIP users should pass through our platform. For now we successfully can make and receive calls inside our domain - NAT is handled fine in most cases, instant messages are handled and so on. It looks like that: - Kamailio receives a call from caller - Call is redirected to our platform - Platform redirects call back to Kamailio and it looks for callee. I've tried to implement that logic. It seems to work in some cases, but I encounter one problem. Sometimes Kamailio can not forward a call to the platform via rewritehostport(). It tries to forward SIP request, but nothing happens - only retransmission handler is called. I mean, after call to t_relay(), I see this: Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... SIP requests seem to be correct, and configuration file works ok from time to time. The problem I've described happens when Kamailio receives a call from another domain (for example, from sipbroker). Inside our domain everything works fine and INVITE passes to our platform correctly. I've attached configuration file and debug output. Files were written for and by Opensips, but with Kamailio I encounter the same problem. Is this a problem with configuration file or something else?

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[Kamailio-Users] roadmap to kamailio 3.0.0
by Daniel-Constantin Mierla 29 Oct '09

29 Oct '09

Hello, I am thinking of planning to release Kamailio 3.0.0 next Wednesday or day after (October 28 or 29). With some delays, testing is going fine in my side, still some bits to be done, here is what I collected: - taking in consideration $du updates in branch route - import fixes from sr_3.0 branch - complete core statistics and handling of drop reply I believe is feasible, in the worse case we can package it at respective date and allow a bit more time for testing, but we unfreeze code and can go for new development. I encourage people to start writing migration documents. Let's see other opinions as well. Cheers, Daniel

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Re: [Kamailio-Users] How to take MAC address
by Zahid Mehmood 28 Oct '09

28 Oct '09

I know that the Polycom phones can be configured to add the MAC address in one of the headers (probably the user agent hdr). -- Zahid ----- Original Message ----- From: users-bounces(a)lists.kamailio.org <users-bounces(a)lists.kamailio.org> To: users(a)lists.kamailio.org <users(a)lists.kamailio.org> Sent: Wed Oct 28 02:58:05 2009 Subject: Re: [Kamailio-Users] How to take MAC address On Wednesday 28 October 2009 06:35:28 Chandrakant Solanki wrote: > Hi > > I want to know the MAC address of registered client on kamailio... > > suppose i have 3 Client registered with 192.168.1.2, 192.168.1.3 and > 192.168.1.4 > > from these 3 Client, i am registered successfully with 101, 102, > 103(a)test.com > > But for additional authentication ... i also need MAC address for this > Client. > > Is there any module, method, function or variable to take MAC address or is > there any alternative for the above case. > > > Waiting for favorable reply... You will not have luck outsite "your proxy network" ... I mean ... that if your proxy is on 192.168.0.10/24 and your clients on any other network, you will always get's the same MAC, the mac of the 192.168.0.10/24 gateway. Your only hope is that SOME SIP devices add their MAC as part of some subscribe or register petition. I don't see any security increase on this, because MAC's could be easily spoofed. -- Raúl Alexis Betancor Santana Dimensión Virtual _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

2 1

Re: [Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Daniel-Constantin Mierla 28 Oct '09

28 Oct '09

On 28.10.2009 11:23 Uhr, Chandrakant Solanki wrote: > > > On Wed, Oct 28, 2009 at 3:50 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 28.10.2009 11:14 Uhr, Chandrakant Solanki wrote: > > > > On Wed, Oct 28, 2009 at 3:18 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com> > <mailto:miconda@gmail.com <mailto:miconda@gmail.com>>> wrote: > > > > On 28.10.2009 10:37 Uhr, Chandrakant Solanki wrote: > > Hi > > Call Sequence is ... > > Client A/B register on Kamailio and both are registered > successfully.. > Firewall is stopped. > > Now... Client A call to Client B.. > > Client A => Kamailio => Asterisk => Kamailio => Client B > > Above is call sequence, when call goes from Asterisk to > Kamailio ... then found below error.... > > ERROR:tm:t_forward_nonack: no branch for forwarding > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: > ERROR:tm:w_t_relay: t_forward_nonack failed > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** > OUT route[8] > > what processing are you doing in kamailio for this case? Do > you do > same failure routing to another destination if B does not > answer? > Such error occurs when r-uri is changed in failure route but no > append_branch() was executed. > > Cheers, > Daniel > > > > hi > > where could i put this thing.... > > Here is my config file for kamailio... > > http://pastebin.com/m34e07510 > > Please help me out ... > > > your config is big to troubleshoot it quick and easy. Definitely > you have call forwarding, what version of kamailio are you using? > the header of file says openser 1.2. > > I suggest you use ngrep/wireshark to watch the sip traffic and put > xlog() messages in the config to correlate and get hints about > what happesn. > > Cheers, > Daniel > > PS. Keep cc-ing to mailing list, there are people that can help > better or faster inc ase they use the sip wizard config. > > -- > Daniel-Constantin Mierla > * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin > * http://www.asipto.com/index.php/sip-router-masterclass/ > > > hi > > i m using kamailio-1.5.2-tls... > if you don't keep mailing list in the discussion thread, your messages will be ignored... Daniel

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Re: [Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Daniel-Constantin Mierla 28 Oct '09

28 Oct '09

On 28.10.2009 11:14 Uhr, Chandrakant Solanki wrote: > > > On Wed, Oct 28, 2009 at 3:18 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 28.10.2009 10:37 Uhr, Chandrakant Solanki wrote: > > Hi > > Call Sequence is ... > > Client A/B register on Kamailio and both are registered > successfully.. > Firewall is stopped. > > Now... Client A call to Client B.. > > Client A => Kamailio => Asterisk => Kamailio => Client B > > Above is call sequence, when call goes from Asterisk to > Kamailio ... then found below error.... > > ERROR:tm:t_forward_nonack: no branch for forwarding > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: > ERROR:tm:w_t_relay: t_forward_nonack failed > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** OUT route[8] > > what processing are you doing in kamailio for this case? Do you do > same failure routing to another destination if B does not answer? > Such error occurs when r-uri is changed in failure route but no > append_branch() was executed. > > Cheers, > Daniel > > > > hi > > where could i put this thing.... > > Here is my config file for kamailio... > > http://pastebin.com/m34e07510 > > Please help me out ... > your config is big to troubleshoot it quick and easy. Definitely you have call forwarding, what version of kamailio are you using? the header of file says openser 1.2. I suggest you use ngrep/wireshark to watch the sip traffic and put xlog() messages in the config to correlate and get hints about what happesn. Cheers, Daniel PS. Keep cc-ing to mailing list, there are people that can help better or faster inc ase they use the sip wizard config. -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] kamctl online
by Chandrakant Solanki 28 Oct '09

28 Oct '09

hello I would like to print user@domainname while i execute kamctl online command... Any Idea -- Regards, Chandrakant Solanki

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[Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Chandrakant Solanki 28 Oct '09

28 Oct '09

Hi Call Sequence is ... Client A/B register on Kamailio and both are registered successfully.. Firewall is stopped. Now... Client A call to Client B.. Client A => Kamailio => Asterisk => Kamailio => Client B Above is call sequence, when call goes from Asterisk to Kamailio ... then found below error.... ERROR:tm:t_forward_nonack: no branch for forwarding Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ERROR:tm:w_t_relay: t_forward_nonack failed Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** OUT route[8] -- Regards, Chandrakant Solanki

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Re: [Kamailio-Users] hung dialogs + no BYE
by Daniel-Constantin Mierla 27 Oct '09

27 Oct '09

On 27.10.2009 21:22 Uhr, Anders wrote: > That was exactly the problem Daniel - no BYE was ever sent from the > UA, so that's what we need to fix! > please keep cc-ing the mailing list. You can fix the problem by identifying the sip devices that do not send BYE. You are doing record routing, right? In 1.5, dialog can send BYE at call timeout -- does not help much you, but can close eventual open channels in gateways. The missing BYE happens when call is between two SIP phones? Or between sip phone and pstn gateway/media server? Cheers, Daniel > Thanks!! > > On Tue, Oct 27, 2009 at 4:16 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> On 26.10.2009 16:41 Uhr, Anders wrote: >> >>> Hi, >>> >>> I have two issues, and I think they are connected. The number of >>> Active Dialogs keeps growing - as if some of them are hung. Not all of >>> them, but some of them. At the same time, I have seen that from a >>> specific customer, there is no BYE in the 'acc' table in the >>> accounting. So, my conclusion - no BYE means it's not finished means >>> it's hung... - right? >>> >>> Any ideas where to look? >>> >>> >>> >> if you don't get a BYE in acc table then maybe was not sent and that keeps >> the dialog active. You can set a max time per call -- timeout -- for each >> dialog: >> >> http://kamailio.org/docs/modules/1.5.x/dialog.html >> >> However, is good to identify why BYE is not coming, maybe is a fraud attempt >> or a broken sip device. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >> * http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] newbie question
by Luis Jimenez 27 Oct '09

27 Oct '09

Hello, list. Exactly what happens with a dialog (at accounting level) when i reload kamailio with dialogs in process? Thanks in advance. Luis Jimenez

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[Kamailio-Users] hung dialogs + no BYE
by Anders 27 Oct '09

27 Oct '09

Hi, I have two issues, and I think they are connected. The number of Active Dialogs keeps growing - as if some of them are hung. Not all of them, but some of them. At the same time, I have seen that from a specific customer, there is no BYE in the 'acc' table in the accounting. So, my conclusion - no BYE means it's not finished means it's hung... - right? Any ideas where to look? Thanks!

2 1

[Kamailio-Users] LCR problem with "id" column and match priorities
by jh＠tutpro.com 24 Oct '09

24 Oct '09

ricardo, in situation where you see higher priority valued gateway group selected first, do you know if the lower priority valued group would be used if none of the gateways in the higher priority valued gateway group can serve the request? if it is used, then both exist in the hash table. the list in the hash table is sorted by prefix length and priority. -- juha

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[Kamailio-Users] acc: accounting flag for CANCEL
by Alex Balashov 24 Oct '09

24 Oct '09

If I have acc set to log CANCEL'd calls, do I need to set the accounting flag on handling CANCEL request, since it is a new transaction relative to the initial INVITE, or will it get logged automatically? Thanks, -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

2 1

[Kamailio-Users] basic SIP forwarding with Asterisk
by Jeff Brower 23 Oct '09

23 Oct '09

All- Can we use Asterisk combined with Kamailio as follows: __________ ___________ | | | | SIP ___| |___ SIP ___| Kamailio |___ SIP | | | rtpproxy | | Asterisk | | | | | | | | | RTP ___| |___ RTP ___| DSP card |___ RTP (G711) |__________| (G711) |___________| (G729, G723, GSM-AMR, EVRC) We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and tear-down and/or when media attributes for the call change? We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't support (or at least doesn't support at higher capacity or without going unstable). Thanks. -Jeff

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[Kamailio-Users] Anybody using domainpolicy module?
by Klaus Darilion 23 Oct '09

23 Oct '09

Hi! The domainpolicy draft is obsolete and the module needs adaptions to work with sip-router. Thus, we plan to remove it, as I guess nobody uses it at all. So, if you use this module, please speek up NOW! regards klaus

1 0

[Kamailio-Users] Search for text with a space in it
by CSB 22 Oct '09

22 Oct '09

OpenSER 1.3 When I try to match a phrase with a space in it I don't get a match. I'm looking for "User-Agent: My UA" None of the following seem to work: if (search("User-Agent:\sMy\sUA")) { if (search("User-Agent:[ ]My[ ]UA")) { if (search("User-Agent: My UA")) { Could anyone point out my obvious mistake? Thanks

2 1

[Kamailio-Users] Kamailio ignores Via in replies?
by Alex Hermann 22 Oct '09

22 Oct '09

Hello, while debugging a problem today I encountered a rather strange issue. The client's UA/router was messing up the Via header, with as result that all the IP's in the Via's were replaced by the public IP of the DSL modem. Now the weird part: because of the mangled Via's, the expected result is that the reply would be routed back to the IP of the client. Instead, the replies were still following the "correct" return path through a series of OpenSER/Kamailio proxies. How is this possible? According to rfc3261, a proxy must forward the reply to the next Via in the received _response_. It seems Kamailio is incorrectly keeping the Via from the request and sends the reply to the stored address of the Via from the _request_. Btw, no 'received' parameter is involved. Can anyone with inside knowledge of Kamailio share his/her view on this issue? And possibly, how to fix it. I suspect I can use check_via=1 to drop incoming replies with incorrect via, although the behaviour of check_via is undocumented (what does it do when check fails?). And why is the default value 0 (as has every example config on the web)? There's also reply_to_via, but according to docs that's only for locally generated replies. Also, its use is beyond me, as the correct value is dependent on the src_ip of the request and thus can't possibly be a global option. Alex.

2 1

[Kamailio-Users] SIP message
by Rajesh 22 Oct '09

22 Oct '09

Dear All, I am using a separate Location server.I would like to know which SIP message is used to call the location server.To be more clear, as INVITE messages are used to make calls, I would like to know the similar SIP message which is used to store the user details in the location table..Is that REGISTER message ?...Correct me if iam wrong.. -- Regards, Rajesh

2 1

[Kamailio-Users] Kamailio v1.5.3 released
by Daniel-Constantin Mierla 21 Oct '09

21 Oct '09

A new release in 1.5 series is out. Kamailio 1.5.3 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0, 1.5.1 or 1.5.2 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.5.3/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/1.5.3/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.3/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/1.5.3/ Modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ What is new in 1.5.x release series is summarized in the announcement of v1.5.0: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes Enjoy and stay tunned -- new major release Kamailio 3.0.0 is going to be out very soon, Daniel

1 0

[Kamailio-Users] LCR problem with "id" column and match priorities.
by Ricardo Martinez 21 Oct '09

21 Oct '09

Hello List. I'm having a weird behavior with the LCR module. I have a particular prefix routed as follows (according to the LCR tables) : mysql> select * from lcr where prefix=0251; +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 89 | 1351 | | 51 | 1 | | 127 | 1351 | | 43 | 2 | +-----+--------+----------+--------+----------+ The gw's group are as follows : mysql> select * from gw where grp_id=51; +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ | 100 | GW1.TERMN1 | 51 | XX.XXX.XXX.16 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 101 | GW2.TERMN1 | 51 | XX.XXX.XXX.13 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 102 | GW3.TERMN1 | 51 | XX.XXX.XXX.32 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 103 | GW4.TERMN1 | 51 | XX.XXX.XXX.35 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ 4 rows in set (0.00 sec) mysql> mysql> select * from gw where grp_id=43; +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ | 76 | GW1.TERMN222 | 43 | XX.XXX.XXX.16 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 77 | GW2.TERMN222 | 43 | XX.XXX.XXX.13 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 78 | GW3.TERMN222 | 43 | XX.XXX.XXX.32 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 79 | GW4.TERMN222 | 43 | XX.XXX.XXX.35 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ 4 rows in set (0.00 sec) So, when a call with prefix 1351 arrived at the proxy, the LCR module is changing the RURI for 1343 (matching the 2nd priority for that prefix). This keep happening even if change the priority value 2 for priority value 5. I even changed the priority 1 for priority 0 as follows : +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 89 | 1351 | | 51 | 0 | | 127 | 1351 | | 43 | 1 | +-----+--------+----------+--------+----------+ But the LCR module keep using the "second" priority. After doing some tests I've found two ways to use the "first" priority for the prefix. 1.- Removing the second priority 2.- Re-ordering the "id" column and using correlative values as indicated in the next table : mysql> select * from lcr where prefix=0251 order by id; +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 127 | 1351 | | 51 | 1 | | 128 | 1351 | | 43 | 2 | +-----+--------+----------+--------+----------+ So, when the prefix are with "no correlatives" values in the "id" column seems to be a problem to match the first priority. But when the values are correlative there is no problem. Is this the correct behavior? Maybe I'm missing something? What happens for example if I need to add a third priority to a certain prefix after doing several changes in the LCR table? I need to re-order every time the "id" values to make them correlatives? I really hope that someone could give me a hand here. Ricardo Martinez.-

3 11

[Serusers] question regarding sip protocol
by Sabatella, Michael 21 Oct '09

21 Oct '09

Hi, I am new to the sip protocol. I have been doing packet traces to understand how the protocol works. I am using the ser sip proxy version 0.9.6. I notice that once my phone registers with a proxy I get a 200 ok. Every 20 seconds after that the client sends a packet to the proxy. It is an Request: OPTIONS packet. My proxy responds with a Status:404 Not found. I tried to do some research on this but could not find anything useful. Does this response of a 404 matter? How can you fix this? I have read that the client does this as an application level ping to make sure the proxy is up. If I have 100s of people registering that is a lot of extra traffic. Is there a way to stop this options query or have the proxy respond properly? Thanks Michael Network Services Engineer ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

2 1

[Kamailio-Users] preparing Kamailio 1.5.3
by Daniel-Constantin Mierla 21 Oct '09

21 Oct '09

Hello, Kamailio 1.5.3 will be released today. Please commit anything you have for branch 1.5 on Sourceforge.net svn in the next two hours. Afterwards we will start preparing the release. Cheers, Daniel

1 0

[Kamailio-Users] max channel capacity
by Jeff Brower 21 Oct '09

21 Oct '09

All- On a reasonably fast server, say quad-core, what is the approx maximum number of G711 IP calls when using Kamailio and rtpproxy? What if rtpproxy runs on a second server? No echo can, no packet concealment, etc... just G711. -Jeff

3 5

Re: [Kamailio-Users] use case for settlement free peering
by Daniel-Constantin Mierla 20 Oct '09

20 Oct '09

welcome -- cc-ing mailing list is recommended all the time -- help people to get the conclusion of discussions. On 20.10.2009 20:04 Uhr, A G wrote: > Great. Thanks for the feedback! > > On Mon, Oct 19, 2009 at 3:47 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> Hello, >> >> On 19.10.2009 20:18 Uhr, A G wrote: >> >>> Greetings: >>> >>> I'm looking for advice on a project/proof of concept I'm working on. >>> I would like to create a settlement-free peering fabric for voice >>> traffic between and among some peer institutions in my area. Because >>> this is more of a side-project for cost cutting measure, I'm primarily >>> looking at open source software, though commercial product >>> recommendations would be helpful as well. >>> >>> The organizations I would like to connect have their own PBXs with >>> large blocks of numbers (whole NPA-NXXs), with no number portability >>> in or out. I imagine both at the individual PBXs and peering fabric, >>> the number routing would be static. To put another way, we would >>> manually configure which connections the block of telephone numbers is >>> reachable at. >>> >>> Here is the required ASCII art diagram :) >>> >>> >>> +-------+ >>> | PBX | >>> +-------+ >>> | >>> +-------+ >>> | SBC | >>> +-------+ >>> | >>> | >>> +---+ +---+ .--------. +---+ +---+ >>> | P | | S | / \ | S | | P | >>> | B |--| B |------ ( ???? )----------| B |--| B | >>> | X | | C | \ / | C | | X | >>> +---+ +---+ `---------' +---+ +---+ >>> | >>> | >>> +-------+ >>> | SBC | >>> +-------+ >>> | >>> +-------+ >>> | PBX | >>> +-------+ >>> >>> >>> >>> >>> For scalability reasons, a full mesh of connections between and among >>> the SBCs is not an attractive option. >>> >>> Here's what I think I need: >>> Basic SIP routing >>> TCP, TLS, and UDP support >>> >>> What would be nice to have: >>> IPv6 >>> CDR >>> >>> What is probably not needed: >>> User agent client registration, presence, IM, voice mail >>> >>> I see there are several different open source voice projects. >>> Do you think this is an appropriate use for Kamailio? >>> >>> >> Kamailio can be used in such scenario. It has a lot of features to help you >> routing calls -- see modules such as lcr, carrierroute, dialplan. >> >> If you do heavy tls/tcp, then upcoming Kamailio 3.0 will have lot of >> improvements in this areas. IPv6 is supported as well for core routing. You >> can generate CDRs using acc module and some stored procedures -- see for >> example: >> http://siremis.asipto.com/install-accounting/ >> >> Cheers, >> Daniel >> >> >>> I'm seeking comments on what you would use for this situation. >>> Are there any existing projects along these lines? >>> Is there one project that is better than another for this application? >>> >>> >>> >> -- >> Daniel-Constantin Mierla >> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >> * http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

2 1

Re: [Kamailio-Users] Problem with LCR "ping"
by Alexandr Dubovikov 20 Oct '09

20 Oct '09

Hi Ricardo, you use ping_interval 10 seconds. For 126 gateways it's extrem short . Can you set this param up to 60 seconds and check again ? Wbr, Alexandr P.S. will be nice to see the result from "ngrep -W byline -t -l port 5060", if it possible :-) > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ricardo Martinez > Sent: Thursday, October 15, 2009 4:48 PM > To: users(a)lists.kamailio.org > Subject: [Kamailio-Users] Problem with LCR "ping" > > Hello list. > > I have a question/problem regarding the LCR module. I'm using Kamailio version 1.5.2. The LCR tables have : > > > > gw table : 126 records > > lrc tables : 120 records. > > > > I'm using the "ping" (ping = 1) feature in all my gateways, part of the cfg file is as follows : > > > > modparam("lcr","db_url","mysql://openserro:openserro@localhost/openser") > > modparam("tm", "fr_inv_timer_avp", "$avp(i:704)") > > modparam("lcr", "gw_uri_avp", "$avp(i:709)") > > modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") > > modparam("lcr", "ruri_user_avp", "$avp(i:500)") > > modparam("lcr", "flags_avp", "$avp(i:712)") > > modparam("lcr", "fetch_rows", 3000) > > modparam("lcr", "ping_interval", 10) > > modparam("lcr", "ping_from", "sip:pinger@mydomain.net") > > modparam("lcr", "ping_method", "OPTIONS") > > > > From time to time I can see 3 o 4 gateways marked as down (with the command kamctl lcr dump). I'm not manually monitoring the gateways every minute so is possible that in a particular moment the gateways could be unreachable, but when I run the command "kamctl lcr dump" the gateways are on-line, that's for sure. > > So I'm not understanding why LCR keep marking the gateways as down, even if they are answering to the OPTIONS request. Could this be a bug in the LCR module? Or maybe I'm missing something?. The silly part of all this is the gateways marked as down are always the same. Could this be related to a memory issue?? > > > > Can someone guide me here?. > > > > Thanks in advance, > > Regards, > > Ricardo Martinez.- > > > > -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info

5 8

[Serusers] SER Staring Problem
by sourav.das1＠wipro.com 20 Oct '09

20 Oct '09

Hi, I am trying to install the SER through the following steps. Download and extract the package : * Download the source package containing mysql support for SER from: http://ftp.iptel.org/pub/ser/latest/ <http://ftp.iptel.org/pub/ser/latest/> * Save the downloaded package at /usr/src folder * Extract the package * Change directory to the extracted package Make and install Make the program and the modules and then install the program in the folder /usr/local. # make && make modules # make prefix=/usr/local install After installing the SER on the Linux system I am trying to start SER through the following command. /usr/local/sbin/serctl start While executing the above command I am getting the following error. Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed Please let me know how to resolve this or something is missing. Regards, Sourav Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

[Kamailio-Users] kamailio 1.5.3 scheduled for next wednesday
by Henning Westerholt 20 Oct '09

20 Oct '09

Hi all, as it has been three month since the last minor release of the kamailio 1.5 branch i'd suggest that we do a 1.5.3 release soon. I want to propose next wednesday the 14.10.2009 as date for the release. So if you have any pending patches in the trunk or in your internal repositories that you want to like to see in this release, please commit them until 12:00 UTC this day. Thank you, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

4 8

[Kamailio-Users] use case for settlement free peering
by A G 19 Oct '09

19 Oct '09

Greetings: I'm looking for advice on a project/proof of concept I'm working on. I would like to create a settlement-free peering fabric for voice traffic between and among some peer institutions in my area. Because this is more of a side-project for cost cutting measure, I'm primarily looking at open source software, though commercial product recommendations would be helpful as well. The organizations I would like to connect have their own PBXs with large blocks of numbers (whole NPA-NXXs), with no number portability in or out. I imagine both at the individual PBXs and peering fabric, the number routing would be static. To put another way, we would manually configure which connections the block of telephone numbers is reachable at. Here is the required ASCII art diagram :) +-------+ | PBX | +-------+ | +-------+ | SBC | +-------+ | | +---+ +---+ .--------. +---+ +---+ | P | | S | / \ | S | | P | | B |--| B |------ ( ???? )----------| B |--| B | | X | | C | \ / | C | | X | +---+ +---+ `---------' +---+ +---+ | | +-------+ | SBC | +-------+ | +-------+ | PBX | +-------+ For scalability reasons, a full mesh of connections between and among the SBCs is not an attractive option. Here's what I think I need: Basic SIP routing TCP, TLS, and UDP support What would be nice to have: IPv6 CDR What is probably not needed: User agent client registration, presence, IM, voice mail I see there are several different open source voice projects. Do you think this is an appropriate use for Kamailio? I'm seeking comments on what you would use for this situation. Are there any existing projects along these lines? Is there one project that is better than another for this application? Thank you

2 1

Presence gateway between SIP and XMPP
by afshin afzali 18 Oct '09

18 Oct '09

Hi Guys, You stated : "*This module is a gateway for presence between SIP and XMPP. * * It translates one format into another and uses xmpp, pua and presence modules to manage the transmition of presence state information*" My first question is that does it enable exchange of presence data between SIP and XMPP in two ways ? If yes how I could use the pua_xmpp_notify and pua_xmpp_req_winfo APIs to do so ? Regards, -- Afshin

1 0

[Kamailio-Users] Odd behavior svn rev 2:5901M
by brandon＠cryy.com 16 Oct '09

16 Oct '09

Hello, I am noticing very rarely however it is happening where kamailio will route to the completely wrong sip uri. In this scenario I see it routing to one of our voicemail applications, thus placing voicemails into the wrong mailbox identifier. What is odd in this particular scenario I was able to capture is that it attempts to deliver for the same call-id to the right location and wrong location within a very very very small time frame causing our voicemail application to pick the wrong branch. What is even odder is this is a recovery from a 487 request cancelled so it is coming from failure route, appending a branch, so there should only be a single branch to voicemail. I clearly see in the logs after setting the ruri that it is correct, I have no idea where it is getting this ghost mailbox uri identifier, it is almost as some how the call got mixed up with a completely different call. Just throwing this out here to see if anyone else has experienced this. I will attach supporting evidence to this email for this scenario shortly. The last thing to mention for now is that it is definitely random and not sure-proof reproducible. Thank you for your time and input ahead of time. Look forward to hearing back from you all! Have a great weekend. Sent from my Verizon Wireless BlackBerry

1 0

[Kamailio-Users] Problem with LCR "ping"
by Ricardo Martinez 16 Oct '09

16 Oct '09

Hello list. I have a question/problem regarding the LCR module. I'm using Kamailio version 1.5.2. The LCR tables have : gw table : 126 records lrc tables : 120 records. I'm using the "ping" (ping = 1) feature in all my gateways, part of the cfg file is as follows : modparam("lcr","db_url","mysql://openserro:openserro@localhost/openser") modparam("tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "fetch_rows", 3000) modparam("lcr", "ping_interval", 10) modparam("lcr", "ping_from", "sip:pinger@mydomain.net") modparam("lcr", "ping_method", "OPTIONS") >From time to time I can see 3 o 4 gateways marked as down (with the command kamctl lcr dump). I'm not manually monitoring the gateways every minute so is possible that in a particular moment the gateways could be unreachable, but when I run the command "kamctl lcr dump" the gateways are on-line, that's for sure. So I'm not understanding why LCR keep marking the gateways as down, even if they are answering to the OPTIONS request. Could this be a bug in the LCR module? Or maybe I'm missing something?. The silly part of all this is the gateways marked as down are always the same. Could this be related to a memory issue?? Can someone guide me here?. Thanks in advance, Regards, Ricardo Martinez.-

4 5

[Kamailio-Users] dispatcher failover between two groups
by Robert McGilvray 16 Oct '09

16 Oct '09

Kam 1.5 I'm using dialplan module to return a setid for the matching regex which is passed onto dispatcher to select the proper gateways. This works fine including failover to the next gateway in the group in failure_route. What I can't quite wrap my head around is how I can setup failover between different dispatcher groups. The scenario is as follows: 2 external PSTN gateways (sip trunks) in group 1 6 internal gateways with PSTN access via PRIs. Two gateways per group for a total of three separate setids in dispatcher. I'm doing some LCR across the 6 internal gateways which are all in different cities via regex matching in dialplan. What I'd like to do is try one of the internal groups first if it matches the local NPA/NXX, and if ALL gateways in that group fail move on to the external gateway group, and try all those gateways (and vice versa in my external gateways fail for non matching numbers). I thought of two possible solutions but I'm not sure if either will work how I expect. 1) Use the priority field in dispatcher to setup a large gateway group that has all the gateways I want to use. - I tested this out in kam 1.5 and the priority field seems to be ignored when hasing on call-id. - What is the intended usage of priority? Will setting two gateways to 0 and the rest to 10 cause the load balancing functions to only select the ones with 0? It seems to be ignoring priority and using all 8 gateways 2) Once in a failure_route can I move the processing into a request_route and call ds_select_dst again with a different setid and append it as a branch? Is there a cleaner solution? Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

1 0

[Kamailio-Users] Codecs
by Anders 16 Oct '09

16 Oct '09

Hi, A very basis 101-question, which I couldn't find the answer to elsewhere: how/where do I see/set codecs allowed and disallowed in Kamailio? Thanks!

2 1

Re: [Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Örn Arnarson 15 Oct '09

15 Oct '09

Hi, I have exactly the same scenario; no NAT involved. Just trying to bridge between these two interfaces. Now, to clarify better what I have done; As I understand it, force_rtp_proxy / rtpproxy_offer needs to be used to inform RTP proxy that it will have to relay the packets. So, that pretty much makes it necessary to use that call. As I understand it, you can't call both fix_nated_sdp and force_rtp_proxy, as the SDP will be mangled (two IP addresses, one after the other). However, I can't get the SDP rewritten properly without using fix_nated_sdp. No matter how I call force_rtp_proxy, I can't specify my own IP address. It works as expected when using fix_nated_sdp though. Here's how I call it: force_rtp_proxy("coe","10.252.1.8"); I've tried multiple variants of flags, c, co, o, coe, coi, i, no flags, etc etc, but no matter what I do, the public interface IP address does not change to the one supplied in the function, and remains the public IP address. So, to summarize; I can change all the headers to be correct with the fix_nated_sdp function. When using that function, RTPProxy does not relay the RTP streams. When I use force_rtp_proxy / rtpproxy_offer, RTPProxy does indeed relay the streams, but does not alter the SDP IP address information to the required specifications. Is there something that I'm missing in the usage of the force_rtp_proxy function, or is it not working as documented? Best regards, Örn 2009/10/14 Alex Balashov <abalashov(a)evaristesys.com>: > Örn Arnarson wrote: > >> Oh, I forgot one thing; >> >> I'm using fix_nated_sdp, as previously mentioned. I have a hunch that >> because I'm not using force_rtp_proxy/rtpproxy_offer it's not being >> relayed (a hunch, because I don't fully understand how rtpproxy >> works). Problem is though, I can't get the IP addresses correct by >> using rtpproxy_offer. > > On behalf of Joe (same company) I will add that there is no NAT scenario > here. Both the private and public network are directly routable from the > rtpproxy box; there is no IP address oversubscription, and no reason to > ignore ports or addresses in SDP and replace them with received parameters. > This is not a far-end NAT traversal solution. > > What is basically being asked here is how to bridge between two interfaces > (internal, external) without NAT using rtpproxy, and if anyone can provide a > canonical implementation or example. > > We've got it running in bridge mode, e.g. > > rtpproxy -F -s udp:10.x.x.x:5050 -l 10.x.x.x/89.y.y.y > > ... and basically, are just wondering how to get this to work on the > Kamailio side using rtpproxy_offer()/rtpproxy_answer(). > > The RTP proxy machine is a dedicated host; the SIP proxies are virtual. > > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 >

1 0

[Kamailio-Users] SIP-T messages
by Luis Jimenez 15 Oct '09

15 Oct '09

Hi, can some one please guide me on how can i parse the received SIP-T messages from the SDP in Kamailio. Thanks in advance Luis Jimenez

2 1

[Kamailio-Users] How to remove route header field
by Mythili D 14 Oct '09

14 Oct '09

hii..Good morning. When we are making a call from one UA to other UA, it passes through proxies. ACK message also passes through the proxies. But i need the ACK message not to go through proxies. It must directly go from UA to UA. How can i make in cfg. There must not be route header filed in SIP meessage flow. Please help me. Thanks & Regards, Mythili.K

2 1

[Kamailio-Users] AVP problem
by Ricardo Martinez 13 Oct '09

13 Oct '09

Hello list. I'm getting this error trying to start the kamailio process /usr/local/sbin/kamailio[29243]: INFO:core:init_mod: initializing module mediaproxy /usr/local/sbin/kamailio[29243]: ERROR:core:pv_parse_spec: error searching pvar "avp" /usr/local/sbin/kamailio[29243]: ERROR:core:pv_parse_spec: wrong char [s/115] in [$avp(s:signaling_ip)] at [5 (5)] /usr/local/sbin/kamailio[29243]: CRITICAL:mediaproxy:mod_init: invalid AVP specification for signaling_ip_avp: `$avp(s:signaling_ip)' /usr/local/sbin/kamailio[29243]: ERROR:core:init_mod: failed to initialize module mediaproxy This is the line in the kamailio.cfg file : modparam("mediaproxy", "signaling_ip_avp", "$avp(s:signaling_ip)") I'm using kamailio 1.5.2 What I'm missing here?. Thanks, Ricardo Martinez

2 2

[Kamailio-Users] SIMPLE vs XMPP: The Resolution
by Iñaki Baz Castillo 13 Oct '09

13 Oct '09

Architectures of XMPP and SIMPLE (OMA specifications) for a full rich presence environment: - XMPP: http://oversip.net/public/XMPP.jpeg - SIMPLE (OMA): http://oversip.net/public/SIP_SIMPLE_OMA.jpeg -- Iñaki Baz Castillo <ibc(a)aliax.net>

8 19

[Kamailio-Users] SIMPLE vs XMPP: The Resolution
by Iñaki Baz Castillo 13 Oct '09

13 Oct '09

Attached the architecture of XMPP and SIMPLE (OMA specifications) for a presence environment. Self explanatory. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 7

[Kamailio-Users] Connecting two or more proxies
by Mythili D 13 Oct '09

13 Oct '09

Hi Good morning. I'm going to connect two proxies. I'm having two servers with openser installed. One user (UAC) will register with one server in one system which is not having openser and the other user (UAS) will register with second server in other syatem which is not having openser. I want to connect the two proxies through cfg file. Hoiw to give the connection. My IP's are UAC registered with 1000(a)192.168.64.77 at other system (192.168.64.85) 192.168.64.77 is one proxy. 192.168.64.94 is secind proxy. UAS registerd with 10(a)192.168.64.94 at other system (192.168.64.128) Now the person at 85 registered with 77 is going to call the person at 128 registered with 94. I have configuered my cfg's in this way: In 77 cfg file, i configuered *if(is_method("INVITE") && !src_ip=="192.168.64.77") * *{ * * rewritehost("192.168.64.77"); * *} * ** In 99 cfg file, i configuered *if(is_method("INVITE") && !src_ip=="192.168.64.94") * *{ * *rewritehost("192.168.64.94"); * *}* Is there any other way to connect. Please help me as soon as possible. Thanks & Regards, Mythili.K

2 1

[Kamailio-Users] keep tos value from incoming SIP message
by kévin l'heveder 13 Oct '09

13 Oct '09

Hi all, I found out how to set the tos value in a outgoing SIP message with the tos parameter from core-cookbook. But my issue is to forward an incoming SIP message keeping the tos value unchange. Is there already a way to do that, something already implemented? Or is there a way to get the tos value from incoming message and dynamically change the tos parameter? Regards,

2 1

[Kamailio-Users] using a proxy and a registrar(only) server to make a cal
by Rajesh 13 Oct '09

13 Oct '09

Dear All, I would like to establish a set up where i have two servers, one acting as a proxy and the other acting as a registrar server(only). Client A ---->Proxy--->Registrar. I am able to register using the above mentioned set up, but am unable to make call.. i get "timed out error" when i try calling. am ataching my cfg below.Any help is greatly appreciated. (My proxy server wil not do registration and my registrar server wil only do registration) ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "imc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #-------tm params----- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) #-------imc params----- modparam("imc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("imc", "hash_size", 8) modparam("imc", "imc_cmd_start_char", "#") #modparam("imc", "imc_cmd_start_char", "#") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication xlog("__The next step is Registration__\n"); rewritehost("192.168.64.77"); xlog("__Registration Successful__\n"); #nr route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain #nr if( is_method("PUBLISH|SUBSCRIBE")) #nr route(2); #nr if (is_method("REGISTER")) #nr { #nr if (!save("location")) #nr sl_reply_error(); #nr exit; #nr } ########## if(uri=~"sip:q.*@") # if(uri=~"sip:q@192.168.64.75") { # IMC - message xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("192.168.64.75", "subscriber")) { www_challenge("192.168.64.75", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { #imc#if (!proxy_authorize("", "subscriber")) { #imc#proxy_challenge("", "0"); #imc#exit; #imc#} #imc#if (is_method("PUBLISH")) #imc#{ #imc# if ($au!=$tU) { #imc# sl_send_reply("403","Forbidden auth ID"); #imc# exit; #imc# } #imc#} else { #imc# if ($au!=$fU) { #imc# sl_send_reply("403","Forbidden auth ID"); #imc# exit; #imc# } #imc#} #imc# consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } -- Regards, Rajesh

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Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Henning Westerholt 13 Oct '09

13 Oct '09

On Montag, 12. Oktober 2009, mashudi wrote: > Hi Henning Westerholt , > did you ever handle SIP-T using Kamailio? > could you share it? Hi Mashudi, sorry, i did not handled SIP-T so far with openser/kamailio. Regards, Henning > > On Friday 13 February 2009, Ovidiu Sas wrote: > >> The multipart body content parsing is embedded in the sdp parser. > >> The sdp parser is available in the source tree under the ./parser/sdp > >> directory. > > > > There is also some documentation about the parser on our wiki: > > > > http://www.kamailio.org/dokuwiki/doku.php/development:sdp-parsing > > > > Cheers, > > > > Henning > > > > _______________________________________________ > > Kamailio (OpenSER) - Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > ***************************************** > > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > > Yogya > > ***************************************** > > ================================= > Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, > DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? > Ikuti Dahsyatnya FLEXI KOMUNITAS. > Ketik CREATE<spasi>[NAMA GRUP], sms ke 345. > Contoh: CREATE SMU2, sms ke 345. > Informasi selanjutnya : > - hubungi 147 > - http://www.telkomflexi.com > - ketik INFO, sms ke 345 > > > ================================= > This Email has been scanned : Anti Virus > =================================

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[Kamailio-Users] Sip-T
by ljjimenez＠gmail.com 12 Oct '09

12 Oct '09

Hi list, can somebody guide on how to read, if it is possible, application/isup payload sent using SIP-T to do routing decisions based of the contents. Thanks in advance Luis Jimenez Luis Jimenez

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[Serusers] Via: branch tags for CANCELs sometimes don't match the one in the INVITE
by Frank Durda IV 12 Oct '09

12 Oct '09

I have been tracking a problem for some time regarding CANCELs that the switch beyond SER was randomly ignoring. There appear to be a number of issues, but one that was identified is caused by SER unexpectedly emitting a branch tag on the Via: header of the CANCEL with a value that isn't identical to the value that appeared in the INVITE message. When the CANCEL arrives at the destination switch with a different branch tag than what was shown in the INVITE, the destination switch correctly rejects the CANCEL with a 481, saying it never heard of this call. So the call doesn't really end. They say they are following RFC 3261, and that does state that the Via: header MUST exactly match. This variance of the branch tag that SER emits doesn't happen every time and is infrequent and random, occurring, perhaps 1% of the time. 99% of the time, the same branch value appears in both INVITE and CANCEL messages. This occasional mismatch is being seen in 2.0.0-rc1 on FreeBSD, Server: SER (2.0.0-rc1 (x86_64/freebsd)) syn_branch=0 is set because branches of zero in ACKs were a really bad thing too. Based on a number of these that have been reviewed, the branch tag generated by SER has three parts, eg branch=STRING1.STRING2.0 I note that STRING1 remains the same but when the malfunction occurs, STRING2 is completely different, as in: FROM INVITE branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c11079.0 FROM CANCEL branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd34935.0 Below is a sample call showing one of these events. Any suggestions on what the problem might be or if this is a known bug (hopefully with a patch) are appreciated. - - - - - - - - - - - - - - - - - - Here is a sample call. IP addresses and phone numbers have been altered, but call ID and branch tags are untouched: CALL ARRIVES AT SER U 2009/10/05 23:00:45.616070 216.0.0.1:5060 -> 208.9.10.11:5060 INVITE sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C P-Asserted-Identity: <sip:8881112222@216.0.0.1:5060> To: sip:9615551212@208.9.10.11:5060 Contact: <sip:8881112222@216.0.0.1:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Max-Forwards: 70 Content-Length: 281 Allow: INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 v=0 o=SIPUAC_1.92.1.8 2077471748 1 IN IP4 55.66.77.88 s=call c=IN IP4 55.66.77.88 t=0 0 m=audio 21356 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 SER SENDS TRYING BACK TO CALLING SWITCH U 2009/10/05 23:00:45.626174 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 100 trying -- your call is important to us From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: sip:9615551212@208.9.10.11:5060 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: SER (2.0.0-rc1 (x86_64/freebsd)) Content-Length: 0 SER SENDS CALL ON TO PSTN SWITCH U 2009/10/05 23:00:45.626192 10.100.20.42:5060 -> 10.133.44.55:5060 INVITE sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C P-Asserted-Identity: <sip:8881112222@216.0.0.1:5060> To: sip:9615551212@208.9.10.11:5060 Contact: <sip:8881112222@216.0.0.1:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Max-Forwards: 16 Content-Length: 301 Allow: INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 v=0 o=SIPUAC_1.92.1.8 2077471748 1 IN IP4 55.66.77.88 s=call c=IN IP4 55.66.77.88 t=0 0 m=audio 55878 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 a=nortpproxy:yes CALLING SWITCH TELLS SER IT IS TRYING U 2009/10/05 23:00:45.629000 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 100 Trying Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Content-Length: 0 CALLING PARTY CHANGES MIND - GOES ON-HOOK U 2009/10/05 23:00:50.579059 216.0.0.1:5060 -> 208.9.10.11:5060 CANCEL sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 SER SENDS CANCEL TO PSTN SWITCH, BUT BRANCH TAGS DON'T MATCH: U 2009/10/05 23:00:50.590039 10.100.20.42:5060 -> 10.133.44.55:5060 CANCEL sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Max-Forwards: 16 Content-Length: 0 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd3493 5.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 NOTE BRANCH MISMATCH: FROM INVITE branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c11079.0 FROM CANCEL branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd34935.0 SER SENDS 200 CANCELING BACK TO CALLING SWITCH U 2009/10/05 23:00:50.590054 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 canceling From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=68cf7f6b22161b0d4dc719dbc13e05af-90a6 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: SER (2.0.0-rc1 (x86_64/freebsd)) Content-Length: 0 PSTN SWITCH REJECTS CANCEL, NOT RECOGNIZING THE CALL PSTN NETWORK CONTINUES TO PROCESS CALL DESPITE CALLER NOW BEING ON-HOOK U 2009/10/05 23:00:50.591489 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 481 Call/Transaction Does Not Exist Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+0+b6afb3ac Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd3493 5.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Content-Length: 0 PSTN SWITCH SENDS BACK ACM/183 Session Progress TO SER U 2009/10/05 23:00:52.619487 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 183 Session Progress Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER SENDS 183 SESSION PROGRESS BACK TO CALLING SWITCH U 2009/10/05 23:00:52.629924 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 183 Session Progress Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER RECEIVES 200 OK FROM PSTN SWITCH U 2009/10/05 23:00:58.129670 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER PASSES 200 OK ON TO CALLING SWITCH U 2009/10/05 23:00:58.137724 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no BECAUSE OF NO RESPONSE TO 200 OK, PSTN SWITCH SENDS IT AGAIN U 2009/10/05 23:00:58.629267 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no U 2009/10/05 23:00:58.639701 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no CALLING SWITCH FINALLY SENDS AN ACK TO SER TO GET RETRIES TO STOP. U 2009/10/05 23:00:58.690296 216.0.0.1:5060 -> 208.9.10.11:5060 ACK sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 ACK Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_2 SER FORWARDS ACK TO PSTN SWITCH. PSTN NETWORK THINKS CALL IS ALIVE, VOIP NETWORK KNOWS CALL IS DEAD. U 2009/10/05 23:00:58.690355 10.100.20.42:5060 -> 10.133.44.55:5060 ACK sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 ACK Max-Forwards: 16 Content-Length: 0 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.1fc8626bef702f1428ad6fd4fd2aa7a f.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_2 .... So the called party has answered the call and can't hear anything. Depending on whether a human or automation was called, this call could remain active on the PSTN for some time.

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 12 Oct '09

12 Oct '09

On Montag, 12. Oktober 2009, Vibhor Singhal wrote: > Please find attached the text file that contains the error that I encounter > while installing Kamailio.I could not get any solution for this.Please > guide me how to resolve this problem. Hi Vibhor, just paste the error message in the mail, its easier then this office document.. You need to install the correct mysql development libraries to be able to compile the db_mysql module. For debian you'll find the package names in the documents in the wiki i posted to you i think. And i'd suggest that you use the latest stable version of kamailio "1.5.2" instead of 1.5.0. This way you'll not run into bugs that were already fixed. Regards, Henning

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 12 Oct '09

12 Oct '09

On Montag, 12. Oktober 2009, Vibhor Singhal wrote: > Thanks for the information. Can you please refer any forum or document that > can guide me on how to make Kamailio as Asterisk's frontend. Hi Vibhor, please CC always to the user list. Just use the search machine of your choice. ;) Here are some example posts with more information: http://www.voip-info.org/wiki/view/Asterisk+at+large http://www.kamailio.org/pipermail/users/2008-February/016181.html http://www.nabble.com/Load-Balancing-Asterisk-using-Kamailio-td23250215.html > And in Kamailio's SIP_trace module, will I get SIP message in the raw form > as mentioned below: Hm, never used this module, but i think it will store the messages in the database. But you could modfiy the module, its open source after all.. Regards, Henning

2 1

[Kamailio-Users] Retransmitted SUBSCRIBEs
by catalina oancea 12 Oct '09

12 Oct '09

Hello, If a phone re-sends a presence SUBSCRIBE requests even after 5 or 500 milliseconds, kamailio replied with '400 Bad request' to the second SUBSCRIBE The scenario is: Phone Kamailio SUBSCRIBE ----------> 401 <---------- SUBSCRIBE ----------> 202 <---------- Pause [ 5ms] (or 500 ms, the same thing happens) SUBSCRIBE ----------> (previous SUBSCRIBE retransmitted) 400 Bad request <---------- Is this the expected behavior? Is there any way to get kamailio to recognize the retransmission? This causes problems with Snom phones because they receive 400 reply when they retransmit the SUBSCRIBE, and presence stops working. Thanks Catalina

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[Kamailio-Users] off-topic :: empty user part of Contact header
by kokoska rokoska 12 Oct '09

12 Oct '09

Hello, I have a little bit off-topic question, but - may be - someone could give me an advice :-) One of our users, when replying to INVITE, sends in 200 OK Contact header without user part: SIP/2.0 200 OK CSeq: 102 INVITE Contact: <sip:1.2.3.4:1234> ... instead of: SIP/2.0 200 OK CSeq: 102 INVITE Contact: <sip:user@1.2.3.4:1234> ... and it fools our SIP<->ISUP GW and it generates completely strange R-URI in ACK :-( My questions are: 1. Is it valid Contact (without user part)? 2. What SIP<->ISUP GW should do (mirror Contact without user part to R-URI in ACK, reject/drop 200 OK etc.)? May be I can ask them to change behaviour... 3. What should do our proxy (drop 200 OK, using htable try to repair R-URI in ACK or Contact in 200 OK etc.)? 4. ??? Any suggestion is very appreciated :-) Best regards, kokoska.rokoska

2 2

Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 09 Oct '09

09 Oct '09

On Freitag, 9. Oktober 2009, Vibhor Singhal wrote: > We are developing CTI applications where we need SIP data from PBX such as > SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for > each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY > etc. > > e.g Wrieshark displays SIP packets for activities going on SIP port in the > below mentioned format: > [..] > > For this we are trying Kamailio as Asterisk frontend.Plz tell us about the > installation process and how to make kamailio as asterisk's fromtend Hallo Vibhor, for installation of kamailio take a look to the wiki, for example at: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn In the "installation" section there are also some more informations: http://www.kamailio.org:80/dokuwiki/doku.php How to use kamailio as a frontend to asterisk has been discussed a lot on this list in the past, the easiest way is probably use the "dispatcher" module. Take a look to the docs for this module for more informations. Best regards, Henning

1 0

[Kamailio-Users] Presence issue - pua dialoginfo
by Francisco Javier Lizarán Vilches 09 Oct '09

09 Oct '09

Hi all: I have a problem using Kamailio 1.5.2 as proxy&presence server. User A is subscribed to B's event dialog, and Kamailio handles this subscription. When a call arrives to pstn number C, the proxy resolves it via ENUM and sends the INVITE to B. The issue is that the PUBLISH request uri is C, because pua_dialoginfo module looks into To uri instead of Request uri, so A is not informed about B state. Has anybody any idea to work around this? Thanks: Francisco Javier Lizarán Vilches ---------------------------------------------- BARIK (Grupo Ormazabal) Departamento Técnico Tel.: +34 917 479 900 www.barik.es <http://www.barik.es/> ----------------------------------------------

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Iñaki Baz Castillo 09 Oct '09

09 Oct '09

2009/10/9 Vibhor Singhal <vibhor.singhal(a)coraltele.com>: > We are a telcom company in Noida developing CTI applications. > We are using Asterisk-1.4.and have configured manager.conf file and > connection with the AMI is successfull using username and password. We want > to monitor/fetch SIP packets from TCP port. Please let us know how to take > SIP data from tcp port. This is a list about Kamailio, not about Asterisk.

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Raúl Alexis Betancor Santana 09 Oct '09

09 Oct '09

On Friday 09 October 2009 06:56:51 Vibhor Singhal wrote: > Hi, > > > We are a telcom company in Noida developing CTI applications. > We are using Asterisk-1.4.and have configured manager.conf file and > connection with the AMI is successfull using username and password. We want > to monitor/fetch SIP packets from TCP port. Please let us know how to take > SIP data from tcp port. First of all .. this is not the correct list to ask this kind of questions . Then Asterisk does not support SIP over TCP (not yet). > Our scenario: > > We are developing CTI applications where we need SIP data from PBX such as > SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for > each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY > etc. Asterisk will not give you that information over the AMI, you will need to develop a module for asterisk or modify chan_sip to archive that goal. > e.g Wrieshark displays SIP packets for activities going on SIP port in the > below mentioned format: > After receiving these SIP Packets from Asterisk,we can parse that data > according to our requirements. > Is there any text/data file or database where asterisk stores all the > information in above format? No > If we need to fetch this data at realtime from the port than how can we do > that? You have toons of ways for skying a cat ... you could ... : - Develop your custom sip-cature program, using libpcap for example - Work with ngrep for getting about what you need - Develop a wireshark filter. - ... > For this we are trying Kamailio as Asterisk frontend.Plz tell us about the > installation process and how to make kamailio as asterisk's fromtend > > > Request for an early reply!!! > Regards If you need such kind of work and you don't have time to learn yourself, better to look up for a consultant and pay for that work. -- Raúl Alexis Betancor Santana Dimensión Virtual

2 1

[Kamailio-Users] How to fetch SIP Data from Asterisk?
by Vibhor Singhal 09 Oct '09

09 Oct '09

Hi, We are a telcom company in Noida developing CTI applications. We are using Asterisk-1.4.and have configured manager.conf file and connection with the AMI is successfull using username and password. We want to monitor/fetch SIP packets from TCP port. Please let us know how to take SIP data from tcp port. Our scenario: We are developing CTI applications where we need SIP data from PBX such as SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY etc. e.g Wrieshark displays SIP packets for activities going on SIP port in the below mentioned format: o SIP Packet as displayed on wireshark******************* SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.35:5060;branch=z9hG4bK10e203f8;rport=5060 Contact: <sip:1000000@192.168.3.15:45118> To: <sip:1000@192.168.3.15:45118>;tag=1f0f342c From: "Unknown"<sip:Unknown@192.168.3.35>;tag=as4a585648 Call-ID: 38670a55c46d1235YzY2YWQxZTM0ODk4NTRhZDI5MjY3YjMyYjE4OTlhZjc. CSeq: 102 NOTIFY User-Agent: X-Lite release 1003l stamp 30942 Content-Length: 0 After receiving these SIP Packets from Asterisk,we can parse that data according to our requirements. Is there any text/data file or database where asterisk stores all the information in above format? If we need to fetch this data at realtime from the port than how can we do that? For this we are trying Kamailio as Asterisk frontend.Plz tell us about the installation process and how to make kamailio as asterisk's fromtend Request for an early reply!!! Regards Regards, Vibhor Singhal Software Engineer Mob : +919891380819 " If you fail to plan,you plan to fail" Coral Telecom Limited E2, Sector 63, Noida - 201301, UP, India. F: +91-120-4035888 E: corporate(a)coraltele.com www.coraltele.com <http://www.coraltele.com/index.asp>

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[Kamailio-Users] Thousands of CANCEL messages - problem with my configuration?
by CSB 08 Oct '09

08 Oct '09

We had a very strange situation last night with our proxy (running OpenSer 1.3). We received an INVITE from a UA. This was then cancelled by the UA. This then resulted in OpenSER generating 40000+ cancel messages with our server talking to itself i.e. the toip and fromip are the same in the sip_trace table. Incidentally, this proxy processes thousands of calls a day so I'm not sure what gave rise to this problem but I guess a particular set of events has highlighted a bug in my config file. The other very curious thing is that somehow our other SIP proxy gets in on the act. Cancel message 40000-odd looks like this: CANCEL sip:031234567@domain.com SIP/2.0 Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.5d037b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.2d037b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.67037b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.20037b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.8ff27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.e16dcb3.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.516dcb3.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.3c5dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.05e27b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.b4e27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.365dcb3.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.52e27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.be4dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.33c27b85.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.2d3dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.84a27b85.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.da2dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.002dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.561dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.111dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.4b627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.ca627b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.5a627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.1a627b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.ed0dcb3.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.d9627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.c9627b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.b9627b85.0 Via: SIP/2.0/UDP 210.48.001.001:5060;branch=z9hG4bK519fe576;rport=5060 From: "Customer" <sip:12345678@domain.com>;tag=as02736614 To: <sip:031234567@domain.com> Contact: <sip:6496306317@210.48.001.001> Call-ID: 771d537932ab79225888a341190691c2(a)domain.com CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 41 Content-Length: 0 I have pasted what I think are the relevant parts from the config file. Could anyone advise where the problem might be? route { # ----------------------------------------------------------------- ### SipTrace. # ----------------------------------------------------------------- if ! is_method("OPTIONS|SUBSCRIBE|NOTIFY") { sip_trace(); setflag(28); } # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #If it's an INVITE & client is NATed, #xlog("welcome"); if (method=="INVITE" && client_nat_test("3")) { #Record-route and specify the record-route header explicitly record_route_preset("147.202.001.001:5060;nat=yes"); # insert IP address xlog("invite and nated so record_route_preset"); #If not Nated and not REGISTER then normal record-route } else if (method!="REGISTER") { record_route(); xlog("not register and not nated so record_route"); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); #call end_media_session in case we're proxying media end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #Ensure we are dealing with a re-INVITE. Only connected calls have tag= #entry in the To header. So if it's loose routed and doesn't have totag #and it's an invite or reply then something's wrong xlog("loose_route"); if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); xlog("403 in lr"); return; }; if (method=="INVITE") { if (!allow_trusted()) { xlog("!allow_ trusted for r-uri <$ru>"); if (!proxy_authorize("","subscriber")) { proxy_challenge("domain.com","1"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; #check client is nated or that we've already identified it's nated xlog("Re-invite. rs $rs si $si rm $rm ru $ru tu $tu fu $fu fd $fd rr $rr"); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); xlog("Recipient is nated so setbflag 6 and use mediaproxy: r-uri <$ru>"); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # Processes out-of-dialogue messages i.e. # 1. new dialogue # 2. message not destined for us (but we are relaying or proxying) xlog("Call Type Processing Section"); if (!is_uri_host_local()) { xlog("!is_uri_host_local"); if (is_from_local() || allow_trusted() ) { xlog("is_from_local() || allow_trusted()"); route(5); route(1); } else { sl_send_reply("403", "Forbidden"); xlog("403 in call type processing. rs $rs si $si rm $rm ru $ru tu $tu fu $fu fd $fd rr $rr"); }; return; }; sip_trace(); if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; } else if (method=="SUBSCRIBE") { route(7); return; }; route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- xlog("hit route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); #Call reply_route(1) to intercept response messages heading to the client t_on_reply("1"); #Try to relay the message to its destination if (!t_relay()) { #If it can't and it's an INVITE or ACK end the media proxying log(1,"route[1] !t_relay"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); } else { log(1,"route[1] t_relay"); }; } onreply_route[1] #Handles message that are returned to the sender i.e. response to caller's original request { xlog("hit onreply_route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); if (isbflagset(6)) { xlog("flag(6) is set for $fu . We're currently in onreply_route(1)"); }; if (isbflagset(7)) { xlog("flag(7) is set for $fu . We're currently in onreply_route(1)"); }; if (status=~"(180)|(183)|2[0-9][0-9]") { xlog("status matches for $fu . We're currently in onreply_route(1)"); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #Check the SDP payload length. Assume if we have something we can call mediaproxy if (!search("^Content-Length:[ ]*0")) { log(1,"using media proxy in onreply_route(1)"); use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Any advice appreciated. Cameron

3 3

[Kamailio-Users] BLF and MWI problem on Grandstreams
by David 08 Oct '09

08 Oct '09

All, I upgraded to Kamailio 1.5 and had a lot of problems getting BLF and MWI working. So I will list here some of my notes in case some body else ever has the same issue. Firstly, in my tests I am using a GXP2000, GXP2020 and a SPA962. I also tested with a SNOM360. my issue was the MWI seemed to fail after an hour, and I could not find why. So in searching my config, I found that when the phones were resubscribing to MWI and BLF, it was being caught in this condition : if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(4); } else { sl_send_reply("404","Not here"); exit; } } My subscribe code handling was after this block, I noticed the issue from the 404 the grandstream was getting when trying to subscribe. I therefore added handling for the subscribe packets in this section, and my MWI and BLF seems to be working. else if ( is_method("PUBLISH") || is_method("SUBSCRIBE") ) { route(5); exit ; } Where route(5) does the appropriate handling of these two types of SIP packets. It should be noted, when I speak of MWI I am referring to presence message-summary which is sent via a publish by my media application. When I write BLF, I am referring to Presence Dialog. So when my phones were resubscribing, they are being sent a 404. I tested the Grandstream and the Linksys, and both seem to have the problem resolved. I will test this with the Snom tomorrow and post my results. David

3 4

[Kamailio-Users] Somebody going to Astricon?
by Klaus Darilion 08 Oct '09

08 Oct '09

..so we should hang out for a beer :-)

4 3

[Kamailio-Users] help with custom module
by liran tal 07 Oct '09

07 Oct '09

Hey list, I've written a very simple module, just exporting one function I need which does something simple like checking a file and returns -1 or 1 if the file exist or not. The problem I'm facing is as follows: During call flow (extension1 calling extension2), it seems that the logic of the call flow is passed via that module and hangs the entire call process. And this happens without even calling that single exported function, but rather just by including the module itself via loadmodule. Log from the call flow: ... 8(3827) SIP Reply (status): 8(3827) version: <SIP/2.0> 8(3827) status: <100> 8(3827) reason: <Trying> 8(3827) parse_headers: flags=2 8(3827) Found param type 232, <branch> = <z9hG4bKde5a.32fd75d7.0>; state=9 8(3827) parse_via: next_via 8(3827) Found param type 232, <branch> = <z9hG4bK-d8754z-303e27391e65ac7d-1---d8754z->; state=6 8(3827) Found param type 235, <rport> = <21644>; state=16 8(3827) end of header reached, state=5 8(3827) parse_headers: Via found, flags=2 8(3827) parse_headers: this is the first via 8(3827) After parse_msg... * 8(3827) forward_reply: found module exec_custom, passing reply to it * 8(3827) DEBUG:destroy_avp_list: destroying list (nil) 8(3827) receive_msg: cleaning up ... that forward_reply thing has the module involved there for whatever reason, I've no idea. This is on openser1.0.1 I'd appreciate some feedback Regards, Liran.

2 1

[Kamailio-Users] PostgreSQL TLS support
by Olle E. Johansson 07 Oct '09

07 Oct '09

The db_postgres documentation doesn't say much about anything, really... Is there support for TLS connections and how do I configure it if it exist? Thanks, /O

2 4

[Kamailio-Users] LDAP and TLS
by Olle E. Johansson 07 Oct '09

07 Oct '09

Greetings, fellow Kamailians! Looking at the LDAP module for Kamailio, I see that we can use LDAPS: and get TLS connections. But where are the SSL configurations so I can use a client cert, or verify the server cert? /O

1 0

[Kamailio-Users] use as registrar only server
by Rajesh 07 Oct '09

07 Oct '09

I would like to know what all can be done using openser/kamailio as a registrar server alone. i.e. in the real world scenario how does openser can be useful as a registrar sever alone.I would like to make myself clear by pointing to the below example. i can integrate openser and asterisk and make openser as a registrar (only)server and can make use of asterisk for call functionality purposes.. Similarly, i would like to know where openser fits in (openser should only be used as a registrar server again). Your guidance is greatly appreciated. -- Regards, Rajesh

1 0

[Kamailio-Users] How can I access dialog attributes from within timeout route?
by Nathan Angelacos 07 Oct '09

07 Oct '09

Hi, In kamailio 1.5.2-notls, I'm trying to get the CallID of a dialog that has timed out: if (is_method( "INVITE" ) && !has_totag()) { $dlg_ctx(timeout_route) = 21; } ... route[21] { xlog ("L_WARN", "Dead dialog: $dlg(callid)"); exit; } However, syslog says: daemon.warn /usr/sbin/kamailio[6578]: Dead Dialog: <null> daemon.warn /usr/sbin/kamailio[6578]: WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID \ '8b2eb5e3-20493c20-14e681d(a)192.168.1.1' and tags '87F489DA-922DF157' '2XrwB5OcEJ2zQvyiTUIh-l2Zd1wjL51M' dlg_handlers.c leads me to believe that the context is loaded before calling the route(21), and the dialog timeout handler is nice enough to report it in syslog just after my script fails. Am I missing something fundamental, or is this just not possible? Thanks!

2 8

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