- sr-users - kamailio.org

(no subject)
by sylver_b 02 Apr '12

02 Apr '12

<a href="http://sxsolar.cl/wp-content/themes/alabastroswp/js/cache/02efpk.html"> http://sxsolar.cl/wp-content/themes/alabastroswp/js/cache/02efpk.html</a>

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(no subject)
by anthony thomas 02 Apr '12

02 Apr '12

<a href="http://cahir.net/roadtrip/journal/archives/02efpk.html"> http://cahir.net/roadtrip/journal/archives/02efpk.html</a>

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CANCEL message processing
by Stephen Dodge (Bistech) 02 Apr '12

02 Apr '12

Hi Guys, I have a question regarding the correct processing for cancelled requests 1. (A)--------------------INVITE ------------------->(B) 2. (A)<------------------180 RIGING------------------(B) 3. (A)--------------------CANCEL-------------------->(B) 4. (A)<------------------OK--------------------------(B) 5. (A)<---------487 Request Terminated---------------(B) 6. (A)--------------------ACK----------------------->(B) I have the following section to deal with cancel messages however I am finding that the 487 response still hits my failure_route block. The failure_route block is called for serial forking. # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) xdbg("## BISNET ## - Cancel Processing\n"); t_relay(); exit; } I can obviously do something like a (!t_check_status("487")) within the failure block but is there a better way to deal with this response ? Although this doesn't really cause me a problem at the moment I see the following error in the debug /usr/sbin/kamailio[23786]: ERROR: tm [tm.c:1368]: ERROR: w_t_relay_to: t_relay_to failed /usr/sbin/kamailio[23786]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: transaction canceled (487/SL) As always your help is appreciated. Steve ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

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uac floods the syslog
by Karsten Horsmann 02 Apr '12

02 Apr '12

Hi! i use kamailio 3.2.2 on Centos 6 and got in my logfile massiv these entries: ERROR: uac [uac.c:462]: src pv value is not string What is the srv pv? Must it be set? -- Mit freundlichen Grüßen *Karsten Horsmann*

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Planning for next major release v3.3.0
by Daniel-Constantin Mierla 02 Apr '12

02 Apr '12

Hello, based on last IRC devel meeting, the next major release was planned to happen before the summer holidays. I am considering freezing the development for v3.3.0 by April 23, test for 4-6 months and release by end of May/beginning of June. If anyone else prefer a different schedule, reply with alternative options. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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Failed to open TLS connection from the client...
by Krishna Kurapati 01 Apr '12

01 Apr '12

I restarted Kamailio after a while and I see this issue... Not sure what triggered this. It was working fine before. Krish Kura Apr 1 07:41:02 (none) /usr/sbin/kamailio[24074]: INFO: tls [tls_domain.c:235]: TLSs<10.1.0.4:5061>: verify_certificate=1 Apr 1 07:41:02 (none) /usr/sbin/kamailio[24074]: INFO: tls [tls_domain.c:238]: TLSs<10.1.0.4:5061>: verify_depth=3 Apr 1 07:41:02 (none) /usr/sbin/kamailio[24074]: INFO: tls [tls_domain.c:567]: TLSs<10.1.0.4:5061>: IF client provides certificate then it MUST be valid Apr 1 07:41:02 (none) /usr/sbin/kamailio[24082]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Apr 1 07:41:41 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3575]: BUG: handle_ser_child: fd -1 for 0 (pid 24074) Apr 1 07:41:41 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3338]: BUG: handle_tcp_child: fd -1061109568 for 0 (pid 0, ser no 0) Apr 1 07:41:41 (none) /usr/sbin/kamailio[24084]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 19 on -1061109568: Bad file descriptor (9) Apr 1 07:41:41 (none) /usr/sbin/kamailio[24084]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0x7ff3c82b3bb8 (flags 0x4018), fd 19 Apr 1 07:42:21 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3575]: BUG: handle_ser_child: fd -1 for 0 (pid 24074) Apr 1 07:42:21 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3338]: BUG: handle_tcp_child: fd -1061109568 for 0 (pid 0, ser no 0) Apr 1 07:42:21 (none) /usr/sbin/kamailio[24084]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 19 on -1061109568: Bad file descriptor (9) Apr 1 07:42:21 (none) /usr/sbin/kamailio[24084]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0x7ff3c82b3bb8 (flags 0x4018), fd 19 Apr 1 07:42:21 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3575]: BUG: handle_ser_child: fd -1 for 0 (pid 24074) Apr 1 07:42:21 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3338]: BUG: handle_tcp_child: fd -1061109568 for 0 (pid 0, ser no 0) Apr 1 07:42:21 (none) /usr/sbin/kamailio[24084]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 19 on -1061109568: Bad file descriptor (9) Apr 1 07:42:21 (none) /usr/sbin/kamailio[24084]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0x7ff3c82b3bb8 (flags 0x4018), fd 19 Apr 1 07:42:45 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3575]: BUG: handle_ser_child: fd -1 for 0 (pid 24074) Apr 1 07:42:45 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3338]: BUG: handle_tcp_child: fd -1061109568 for 0 (pid 0, ser no 0) Apr 1 07:42:45 (none) /usr/sbin/kamailio[24084]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 19 on -1061109568: Bad file descriptor (9) Apr 1 07:42:45 (none) /usr/sbin/kamailio[24084]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0x7ff3c82b3bb8 (flags 0x4018), fd 19 Apr 1 07:43:39 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3575]: BUG: handle_ser_child: fd -1 for 0 (pid 24074) Apr 1 07:43:39 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3338]: BUG: handle_tcp_child: fd -1061109568 for 0 (pid 0, ser no 0) Apr 1 07:43:39 (none) /usr/sbin/kamailio[24084]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 19 on -1061109568: Bad file descriptor (9) Apr 1 07:43:39 (none) /usr/sbin/kamailio[24084]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0x7ff3c82b3bb8 (flags 0x4018), fd 19 Apr 1 07:45:22 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3575]: BUG: handle_ser_child: fd -1 for 0 (pid 24074) Apr 1 07:45:22 (none) /usr/sbin/kamailio[24084]: : <core> [tcp_main.c:3338]: BUG: handle_tcp_child: fd -1061109568 for 0 (pid 0, ser no 0) Apr 1 07:45:22 (none) /usr/sbin/kamailio[24084]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 19 on -1061109568: Bad file descriptor (9) Apr 1 07:45:22 (none) /usr/sbin/kamailio[24084]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0x7ff3c82b3bb8 (flags 0x4018), fd 19

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need advice
by Nima Jalali 01 Apr '12

01 Apr '12

*Hello dear iptel supporter * Sorry, Is pbxes.org yours ? if No : I need a one sip trunk(with many extensions) with my domain hosted. if Yes : How can i hosted my domain on pbxes ? Please Advice . I Appreciate it . Sincerely Yours Nima Jalali

1 0

siptrace error...
by Krishna Kurapati 30 Mar '12

30 Mar '12

Hi, I keep getting this error in the sip_trace module: Mar 26 06:19:26 siptest /usr/sbin/kamailio[4279]: ERROR: siptrace [siptrace.c:1705]: there is not a valid number port 5060?/?? Krish Kura

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SEMS IVR issue
by Uri Shacked 30 Mar '12

30 Mar '12

Hi, If anyone here is familiar with SEMS.... I installed it today, and could not figure out how to load the IVR application... there is no ivr.so file.... any ideas? there is so little info about it on the web... BR, Uri

3 2

acc log in two databases
by Johannes Wagner-Meingassner 30 Mar '12

30 Mar '12

hi, is it possible configure accounting to log into 2 databases at the same time? thanks, hannes modparam("acc", "db_url", "mysql://user:pw@localhost/ser") modparam("acc", "db_url", "mysql://user:pw@10.10.10.10/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3)

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Kamailio and billing services .
by Ryan Gholam 30 Mar '12

30 Mar '12

hello, Is there any billing application or modules to use with kamailio , for example if i want to check the credits of the user on every passing minute , and when he's out of credits i stop the call . Please reply ASAP . Thank you

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Kamailio Shutdown when received a packet without RR
by Javier Vidal 30 Mar '12

30 Mar '12

Hi all, when i send a packet to a carrier with topoh activated he answers a packet without RR and kamailio shutdown himself. I understand that my Carrier shouldn't send me a SIP Packet without RR, but i also can't understand why kamailio shutdowns. Thank you in advance and Good IPv6 Friday. :) DEBUG: Feb 24 14:52:29 sbc /usr/sbin/kamailio[18305]: DEBUG: topoh [th_msg.c:514]: no record route header Feb 24 14:52:29 sbc /usr/sbin/kamailio[18305]: DEBUG: topoh [th_msg.c:906]: +++++++++++++ added cookie header [TH: uch#015#012] Feb 24 14:52:29 sbc /usr/sbin/kamailio[18317]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 24 Feb 24 14:52:29 sbc /usr/sbin/kamailio[18317]: DEBUG: <core> [tcp_main.c:3552]: DBG: handle_ser_child: dead child 10, pid 18305 (shutting down?) Sip Traces: U IPA.IPA.IPA.IPA:5060 -> IPB.IPB.IPB.IPB:5060 INVITE sip:XXXX@XXXX SIP/2.0. Record-Route: <sip:XXXX;lr=on;ftag=ej9ta117t310r;nat=yes;did=01e.e6a1617>. Via: SIP/2.0/UDP 0;branch=z9hG4bK711.2c4d5be1.0. Via: SIP/2.0/UDP 192.168.0.1;branch=z9hG4bKsr-YD8Tc92pXoRWfxhHXugAcu29X-rkcu2ASuyY-jkFlv6zbai0baTRSPhOXjyIbGxp6OHRNu8df9fI-9iOlaU5WPm2EuioEGY*. Max-Forwards: 33. From: "XXXX" <sip:XXXX@XXXX>;tag=ej9ta117t310r. To: <sip:XXXX@XXXX:5060>. Call-ID: !!:XuWFwjmG4Phy4jF16-DKXuUGcP5Iwjby6GX9XjWI4jUIwjh9. CSeq: 24711539 INVITE. Contact: <sip:192.168.0.1;line=sr-bO8JwGmqLkfiPgyfTj2KXIrAX9gpS-rAXu6QSPhOXjyDbGxpbqi0baTRmEfJwOmqsWfiPgyf>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 224. X-FS-Support: update_display. Remote-Party-ID: "XXX" <sip:XXXX@XXXX>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1330077574 1330077575 IN IP4 XXXX. s=FreeSWITCH. c=IN IP4 XXXX. t=0 0. m=audio 38806 RTP/AVP 18 8 0 101 13. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=nortpproxy:yes. U IPB.IPB.IPB.IPB:5060 -> IPA.IPA.IPA.IPA:5060 SIP/2.0 100 Trying. From: "XXXX" <sip:XXXX@XXXX>;tag=ej9ta117t310r. To: <sip:XXXX@XXXX>;tag=1491ef95c94cd6e3fd8ab36b. CSeq: 24711539 INVITE. Call-ID: !!. Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK711.2c4d5be1.0. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKsr-YD8Tc92pXoRWfxhHXugAcu29X-rkcu2ASuyY-jkFlv6zbai0baTRSPhOXjyIbGxp6OHRNu8df9fI-9iOlaU5WPm2EuioEGY*. Content-Length: 0.

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Reload lua script?
by Javier Gallart 30 Mar '12

30 Mar '12

Hello all I'm starting to play with lua scripting in Kamailio. It's really cool, some complex stuff in the script becomes really easy this way. The script is becoming quite big, therefore I'd like to use lua_run instead of lua_dofile. My doubt is pretty basic: if i load a script in memory at startup via the "load" parameter and the script is modifed afterwards, is there any way to reload the module so the script is reread? Thanks in advance Regards Javi

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alphabetic indexes for devel version
by Daniel-Constantin Mierla 30 Mar '12

30 Mar '12

Hello, I generated alphabetic indexes for module parameters, functions, mi commands and statistics: - https://www.kamailio.org/wiki/start#alphabetic_indexes If any of you find errors or missing items, report back to the lists. A note for developers, if you have to use '[[' in the __title__ of the sections for these components, add a space between them otherwise will conflict with wiki syntax and break the indexing (such '[' is commonly used for optional parameters). Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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Re: [SR-Users] how to configure Kamailio in ECHO mode
by Alex Balashov 29 Mar '12

29 Mar '12

No, it's not possible. I think you have the wrong idea about what Kamailio is. Kamailio neither receives nor produces audio at all. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com "Fomitchev, AndreiX" <andreix.fomitchev(a)intel.com> wrote: >Oups, sorry, bad manipulation: >I want to configure Kamailio in echo mode, i.e. SIP server repeats the input audio into the output, so I can hear myself with a very small amount of delay... > >Is it possible? If so, how? > >Thank you for your help. > >Best regards, > >From: Fomitchev, AndreiX >Sent: Thursday, March 29, 2012 1:34 PM >To: 'sr-users(a)lists.sip-router.org' >Subject: how to configure Kamailio in ECHO mode > >Hi, > >For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. SIP server >--------------------------------------------------------------------- >Intel Corporation SAS (French simplified joint stock company) >Registered headquarters: "Les Montalets"- 2, rue de Paris, >92196 Meudon Cedex, France >Registration Number: 302 456 199 R.C.S. NANTERRE >Capital: 4,572,000 Euros > >This e-mail and any attachments may contain confidential material for >the sole use of the intended recipient(s). Any review or distribution >by others is strictly prohibited. If you are not the intended >recipient, please contact the sender and delete all copies. > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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branches usage
by Uri Shacked 29 Mar '12

29 Mar '12

Thanks. so, when i do append_branch and t_relay... i will send two invites at once, right? BR, Uri

2 1

early media: How to change to tag
by Min Wang 29 Mar '12

29 Mar '12

HI I am trying to do the early media like this: PSTN/GW1 -> Kamailio -> Asterisk --> PSTN/GW2 GW1 send invite to Kamilio, Kamilio proxy to Asterisk, Asterisk reply with 183 ( with to tag=A) and sdp After the playing some medium, asterisk send 603 reply to Kamailio Kamilio intercept the 603, and lookup the routing table, decide to send invite to GW2. GW2 reply with 180/200OK with to tag=B In order to properly proxy the msg to GW1, Kamailio seems need to change the to tag from B to A. Is it the right way to do? If so how to change the to tag? or do I need B2BUA ( only signalling) to do the early media? thanks. min

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Update of wiki write access
by Daniel-Constantin Mierla 29 Mar '12

29 Mar '12

Hello, just to let you know that over the weekend the ACL (user authentication) for write access to wiki portals of the project has been turned on. Anyone that wants to write to the wiki has to create herself/himself an account. The reason for requiring user authentication is weakness of the captha system used for anonymous posting -- during the past weeks lot of wiki pages were updated by spam bots. If anyone is aware of a stronger captcha-like system to be used for anonymous posting, write back here. It has to work with dokuwiki. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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Re: [SR-Users] how to configure Kamailio in ECHO mode
by Alex Balashov 29 Mar '12

29 Mar '12

Probably a user agent that is actually an endpoint, like a PBX or media server. You should take a look at Asterisk or Freeswitch. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com "Fomitchev, AndreiX" <andreix.fomitchev(a)intel.com> wrote: >Hi Alex, > >Thank you for your response. Please, could you tell me what I need to setup a SIP “echo service”? I am new in SIP domain :/ > >Regards, > >From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov >Sent: Thursday, March 29, 2012 1:38 PM >To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List >Subject: Re: [SR-Users] how to configure Kamailio in ECHO mode > >No, it's not possible. I think you have the wrong idea about what Kamailio is. Kamailio neither receives nor produces audio at all. > >-- >Alex Balashov - Principal >Evariste Systems LLC >235 E Ponce de Leon Ave >Suite 106 >Atlanta, GA 30030 >Tel: +1-678-954-0670 >Fax: +1-404-961-1892 >Web: http://www.evaristesys.com/, http://www.alexbalashov.com > >"Fomitchev, AndreiX" <andreix.fomitchev(a)intel.com<mailto:andreix.fomitchev@intel.com>> wrote: >Oups, sorry, bad manipulation: >I want to configure Kamailio in echo mode, i.e. SIP server repeats the input audio into the output, so I can hear myself with a very small amount of delay… > >Is it possible? If so, how? > >Thank you for your help. > >Best regards, > >From: Fomitchev, AndreiX >Sent: Thursday, March 29, 2012 1:34 PM >To: 'sr-users(a)lists.sip-router.org' >Subject: how to configure Kamailio in ECHO mode > >Hi, > >For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. SIP server >--------------------------------------------------------------------- >Intel Corporation SAS (French simplified joint stock company) >Registered headquarters: "Les Montalets"- 2, rue de Paris, >92196 Meudon Cedex, France >Registration Number: 302 456 199 R.C.S. NANTERRE >Capital: 4,572,000 Euros > >This e-mail and any attachments may contain confidential material for >the sole use of the intended recipient(s). Any review or distribution >by others is strictly prohibited. If you are not the intended >recipient, please contact the sender and delete all copies. >--------------------------------------------------------------------- >Intel Corporation SAS (French simplified joint stock company) >Registered headquarters: "Les Montalets"- 2, rue de Paris, >92196 Meudon Cedex, France >Registration Number: 302 456 199 R.C.S. NANTERRE >Capital: 4,572,000 Euros > >This e-mail and any attachments may contain confidential material for >the sole use of the intended recipient(s). Any review or distribution >by others is strictly prohibited. If you are not the intended >recipient, please contact the sender and delete all copies. > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] how to configure Kamailio in ECHO mode
by Alex Balashov 29 Mar '12

29 Mar '12

Can you elaborate on what that means? If you are talking about acoustic echo, it's not possible. Kamailio is a proxy / SIP server, it does not handle media or answer calls or anything like that. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com "Fomitchev, AndreiX" <andreix.fomitchev(a)intel.com> wrote: >Hi, > >For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. SIP server >--------------------------------------------------------------------- >Intel Corporation SAS (French simplified joint stock company) >Registered headquarters: "Les Montalets"- 2, rue de Paris, >92196 Meudon Cedex, France >Registration Number: 302 456 199 R.C.S. NANTERRE >Capital: 4,572,000 Euros > >This e-mail and any attachments may contain confidential material for >the sole use of the intended recipient(s). Any review or distribution >by others is strictly prohibited. If you are not the intended >recipient, please contact the sender and delete all copies. > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] how to configure Kamailio in ECHO mode
by Fomitchev, AndreiX 29 Mar '12

29 Mar '12

Oups, sorry, bad manipulation: I want to configure Kamailio in echo mode, i.e. SIP server repeats the input audio into the output, so I can hear myself with a very small amount of delay... Is it possible? If so, how? Thank you for your help. Best regards, From: Fomitchev, AndreiX Sent: Thursday, March 29, 2012 1:34 PM To: 'sr-users(a)lists.sip-router.org' Subject: how to configure Kamailio in ECHO mode Hi, For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. SIP server --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

1 0

how to configure Kamailio in ECHO mode
by Fomitchev, AndreiX 29 Mar '12

29 Mar '12

Hi, For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. SIP server --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

1 0

branches usage
by Uri Shacked 29 Mar '12

29 Mar '12

Hi, Thanks, this was very helpful for understanding. Still let’s see if i got it right: I get the INVITE. Do whatever I do on the main route. Then do: “append_branch();” “changes….changes…..;” Now before the t_relay do t_on_branch(name of branch). Get the reply (301 in my case) And from here I am back to the original request? maybe a stupid question now but - I must use the append_branch() right :-)? BR, Uri Hi, On 03/27/2012 04:46 PM, Uri Shacked wrote: >* In my case i need to change the header and then send it. *>* *>* Here the case works if i make the changes after i sent the invite and *>* got the reply. As i know the t_on_branch will work after the invite is *>* sent... am i wrong? *>* *>* There are many questions to ask about the branches... when do they work *>* in a serial manner, when parallel... where can i find it documented best? * Actually it's quite simple. If a request comes in and you modify it in a normal route (e.g. calling rtpproxy_offer(), uac_replace_from(), adding additional headers and stuff like that) and you send it out using e.g. t_relay(), then in a serial scenario when you re-enter via a failure-route, these changes will still be there (which is not always what you want). On the other hand, if you register a branch route via t_on_branch() and do your changes there instead of prior to calling t_relay(), then these changes from branch route will be rolled back, and you can do your changes easily again in the same or another branch route starting with your original message. This is important for example in serial call forward scenarios, where the first destination might require an rtpproxy, but the second doesn't, or when the first destination requires different custom headers compare to the second. Basically the same applies to parallel scenarios. If you modify your message before calling lookup() and you've multiple contacts registered for your destination, then your changes will apply for all branches which are created by that. However if you move your modifications to a branch route, then you can work on your messages individually, depending on the branch. Hope this clarifies it a bit. Andreas

2 2

AVPs in timer-route
by Klaus Darilion 29 Mar '12

29 Mar '12

Hi! I wonder what happens to AVPs defined in a route which is executed by rtimer? It seems (tests on Kamailio 1.5) that the AVPs, once they are defined, they life forever. This would be fine for my application. Are there any issues when using AVPs in timer routes? Thanks Klaus

2 3

ITSPA 2012 Award for Open Source VoIP Projects
by Daniel-Constantin Mierla 28 Mar '12

28 Mar '12

Hello, ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry: * http://www.itspaawards.org.uk/ Open Source VoIP Projects won a special category this year, Members' Pick, for providing a real value to VoIP Industry. I had the chance to attend the event in London and I have been selected to pick up the award. I made a news on kamailio.org website with more details: * http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projec… As you would expect, a complete voip platform usually involves several open source projects, for components such as load balancers, registrar, proxy or media servers, thus the decision of ITSPA for awarding to the group. I am reiterating the thanks to everyone in the community, developers or users, for their dedication in improving the project. This time extended credits to our British community, very active in both development and usage, for their support in this awards ceremony and overall advertising on the UK markets. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

2 2

TCP memory failures...
by Krishna Kurapati 28 Mar '12

28 Mar '12

I keep getting these... I am using master branch... Mar 26 06:46:34 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:1063]: ERROR: tcpconn_new: mem. allocation failure Mar 26 06:46:34 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:4135]: ERROR: handle_new_connect: tcpconn_new failed, closing socket Mar 26 06:46:48 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:1063]: ERROR: tcpconn_new: mem. allocation failure Mar 26 06:46:48 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:4135]: ERROR: handle_new_connect: tcpconn_new failed, closing socket Mar 26 06:47:51 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:1063]: ERROR: tcpconn_new: mem. allocation failure Mar 26 06:47:51 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:4135]: ERROR: handle_new_connect: tcpconn_new failed, closing socket Mar 26 06:48:45 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:1063]: ERROR: tcpconn_new: mem. allocation failure Mar 26 06:48:45 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:4135]: ERROR: handle_new_connect: tcpconn_new failed, closing socket I have never seen this before... what could cause this? Is there something I could do to fix this issue? Krish Kura

2 3

uac with more then one target?
by Karsten Horsmann 28 Mar '12

28 Mar '12

Hello, my kamailio proxy handles calls between public sip-clients and internal ivr systems. Some customer give us calls via sip-trunks to our proxy and this goes to the ivr too. Now some ask if we can act as UAC to get calls from them, for example authenticate to sipgate and feed the landline calls to our ivr system via kamailio. AFAIK the module UAC provides only one pair of user/password credentials. Is that right? Have i here a chance to do the job with kamailio or must i use an second voip system for that? -- Mit freundlichen Grüßen *Karsten Horsmann*

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Re: [SR-Users] Search all dialogs - Kamailiio 3.0.3
by Daniel-Constantin Mierla 28 Mar '12

28 Mar '12

Hello, On 3/23/12 2:00 PM, Charles Chance wrote: > > Hi all, > > I was wondering if anyone could offer any advice please. > > We are using the dialog module and need to be able to find out whether > another dialog exists within a particular profile and having a > specific value, so that we can send the new call to the same gateway. > As far as I am aware, the only way to see other dialogs from the > request route is to query the database (having set db mode to > realtime). This is ok, but the profile/value is not stored in the > database table. Is there a way of fetching the information that I am > missing, or should we just add some customisation to the module so > that it also writes this information to db? > in the last version you can attach variables to dialog and they are stored in database -- but that requires you to upgrade. Also, devel version saves the profiles in database, in a json format. What you can do with 3.0.x: - use sqlops to write to db table the info you need -- you would have to update db based on call events, such as initial invite: insert record to db; 200ok to initial invite, confirm the dialog in the record; bye: delete the db record; you need to delete also for the case of negative reply to invite. You can use rtimer to delete old records (those that don't get bye). This is pretty much what dialog module does, but hides it in the c code -- all doable in the config - use htable module to store the details in memory instead of a db table In summary: upgrading to 3.2.x makes your life much easier. Next major release comes with a better dialog storage, you may want to play with it as well. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

2 1

siptrace and duplicate_uri - not forwarding the packets
by Mino Haluz 28 Mar '12

28 Mar '12

Hi, I have siptrace working but when I set duplicate_uri, no packet is sent to the destination. Was there some bug fix or anything I have to set further? There is nothing in syslog, siptrace locally is stored fine. modparam("siptrace", "db_url", "mysql://localhost/kamailio") # Database URL # modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) # Flag is used to mark messages to trace modparam("siptrace", "trace_sl_acks", 0) # Do not trace ACKs separately. They are traced during normal sip_trace() modparam("siptrace", "duplicate_uri", "sip:IP1:5060") kamailio 3.2.0 Thanks Mino

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kamailio 3.2.2 undefined symbol: boot_OpenSER while loading module perl
by Karsten Horsmann 28 Mar '12

28 Mar '12

Hi, i am using Centos 6 (32bit) and kamailio 3.2.2 build from the tar.gz. I installed all dependencies for the modules and the make all, make install seems to be fine. But something is wrong with my kamailio - how can i fix this? If i try to load the perl module i get this error messages: ERROR: <core> [sr_module.c:565]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/perl.so>: /usr/local/lib/kamailio/modules_k/perl.so: undefined symbol: boot_OpenSER <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 214, column 12-20: failed to load module ls -la /usr/local/lib/kamailio/modules_k/perl.so -rwxr-xr-x 1 root root 107985 28. Mär 09:38 /usr/local/lib/kamailio/modules_k/perl.so ldd /usr/local/lib/kamailio/modules_k/perl.so linux-gate.so.1 => (0x008fe000) libperl.so => /usr/lib/perl5/CORE/libperl.so (0x00dd8000) libresolv.so.2 => /lib/libresolv.so.2 (0x00343000) libnsl.so.1 => /lib/libnsl.so.1 (0x00cdf000) libdl.so.2 => /lib/libdl.so.2 (0x0024f000) libm.so.6 => /lib/libm.so.6 (0x00fab000) libcrypt.so.1 => /lib/libcrypt.so.1 (0x002c3000) libutil.so.1 => /lib/libutil.so.1 (0x00d21000) libpthread.so.0 => /lib/libpthread.so.0 (0x00110000) libc.so.6 => /lib/libc.so.6 (0x0035c000) libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00aa1000) /lib/ld-linux.so.2 (0x004f7000) libfreebl3.so => /lib/libfreebl3.so (0x00632000) -- Kind Regards Mit freundlichen Grüßen *Karsten Horsmann*

2 1

CRLF Error with tcp_crlf_ping
by Jijo 27 Mar '12

27 Mar '12

Hi All. As per the tcp_crlf_ping description, kamailio shall reply CRLF only when it receives "CRLF CRLF". But In the 3.1.0 version, if kamailio recieves "CRLF", it respond back CRLF. Is this an expected behavior or an error? Thanks Jijo

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Re: [SR-Users] Error loading pv.so undefined symbol: msg_ctx_id_set
by Daniel-Constantin Mierla 27 Mar '12

27 Mar '12

Hello, On 3/27/12 9:33 PM, Simpson Chua wrote: > Hi, > > [..] > > # /usr/local/kamailio/sbin/kamailio -V > version: kamailio 3.3.0-dev6 (x86_64/linux) 6853a8 for devel version be sure you run latest version from master branch. I just started with default config and runs fine, no missing symbol. Cheers, Daniel > [...] > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *To:* Simpson Chua <simpsonchua(a)yahoo.com>; SIP Router - Kamailio > (OpenSER) and SIP Express Router (SER) - Users Mailing List > <sr-users(a)lists.sip-router.org> > *Sent:* Tuesday, March 27, 2012 2:23 PM > *Subject:* Re: [SR-Users] Error loading pv.so undefined symbol: > msg_ctx_id_set > > Hello, > > are you sure you are running 3.2.x? That version does not have > msg_ctx_id_set() at all yet. Can you send the output of 'kamailio -V'? > > Cheers, > Daniel > > On 3/27/12 8:51 PM, Simpson Chua wrote: >> Hi, >> >> I'm getting the following error when attempting to load pv.so >> >> kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could >> not open module </usr/local/lib64/kamailio/modules_k/pv.so>: >> /usr/local/lib64/kamailio/modules_k/pv.so: undefined symbol: >> msg_ctx_id_set >> >> I'm running Kamailio 3.2.x. >> >> Thanks, >> Simpson >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany > http://www.asipto.com/index.php/kamailio-advanced-training/ > > -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Error loading pv.so undefined symbol: msg_ctx_id_set
by Simpson Chua 27 Mar '12

27 Mar '12

Hi, I'm getting the following error when attempting to load pv.so kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules_k/pv.so>: /usr/local/lib64/kamailio/modules_k/pv.so: undefined symbol: msg_ctx_id_set I'm running Kamailio 3.2.x. Thanks, Simpson

2 1

Re: [SR-Users] branches usage
by Uri Shacked 27 Mar '12

27 Mar '12

On Tue, Mar 27, 2012 at 3:13 PM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > > > > Following the advice I got on the subject “remove_hf" and "append_hf" one > after the other , twice, issue”, I tried to work with the “append_branch()” > function. > > What I need to do is, after I received the invite from the original user, > I use SIP methods to work with the NP server. I subst() the user part, send > the call to the NP server, get the reply, parse it and then subst() back to > the destination and send the call out. > > Doing it with “append_branch” seems logical. > > I do “append_branch()” before changing the user, get the reply (301), > goes on to failure route after the branch route was called, and from there > I have a problem. > > I need to continue my flow without the branch I added. How do I return to > the original on? > > I read about the t_drop_replies but didn’t quit understand….. > > > > BR, > > Uri >

4 5

Re: [SR-Users] Kamailio modifying ruri during loose-route - SOLVED
by Dan-Cristian Bogos 27 Mar '12

27 Mar '12

Another one bites the dust ;). My problem was strict vs loose routing done by loose_route function. In order to loop through B2BUA I have defined the loopback interface's IP as a local alias for simplicity in checking myself in routing script. When doing loose_route, the route address was for some reason not matching my server and due to strict routing the RURI was rewritten with the weird address in the route header. Hope my report helps someone in the future. DanB

1 0

"remove_hf" and "append_hf" one after the other , twice, issue.
by Uri Shacked 27 Mar '12

27 Mar '12

Thanks. I read option 2 and i agree on 1 :-) Didnt use any "branch stuff" yet :-(... it seems very logical to do so when i send the call to the NP server. Will hit the docs... Hi, when you are adding the new Contact-Header in Failure-Route (after you received the 3xx reply), you are working on the original message again. So the old Contact is still removed and the new Contact is added (as for the initial request). There are two options for you: 1) Move your remove_hf/append_hf into a branch route (so the changes are only for the current branch) 2) Use "msg_apply_changes()" before removing and adding the header again (see http://kamailio.org/docs/modules/devel/modules/textopsx.html#textopsx.msg_a… ) I i were you, i'd choose option 1, since for Option 2 the message and the headers need to be parsed again.... (performance penalty). Good luck, Carsten 2012/3/27 Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >: >* Hi, *>* *>* *>* *>* In the scenario i am using, the flow is as followed: *>* *>* 1. Getting the INVITE from the user *>* *>* 2. Sending an INVITE with RURI changes to the NP server (sip NP server) *>* *>* 3. Getting a 301 reply (using the reply for the NP code) *>* *>* 4. Sending the call to the destination. *>* *>* *>* *>* My problem is that the NP server needs some parameters in the contact *>* header. *>* *>* So, here is what i do : *>* *>* *>* *>* First, before sending the invite to the NP: *>* *>* *>* *>* remove_hf("Contact"); *>* *>* append_hf("Contact *>* :<sip:XXX at server_ip<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> :port;param1=p1;param2=p2>\r\n","Contact"); *>* *>* *>* *>* (The invite is sent perfectly and works great) *>* *>* *>* *>* After the NP reply i do: *>* *>* *>* *>* remove_hf("Contact"); *>* *>* append_hf("Contact :$ct","Contact"); *>* *>* *>* *>* Now I notice there are two contact headers. It seems that the remove_hf *>* didn’t work this time. *>* *>* Any ideas? *>* *>* *>* *>* Thanks, *>* *>* Uri *

1 0

"remove_hf" and "append_hf" one after the other , twice, issue.
by Uri Shacked 27 Mar '12

27 Mar '12

Hi, In the scenario i am using, the flow is as followed: 1. Getting the INVITE from the user 2. Sending an INVITE with RURI changes to the NP server (sip NP server) 3. Getting a 301 reply (using the reply for the NP code) 4. Sending the call to the destination. My problem is that the NP server needs some parameters in the contact header. So, here is what i do : First, before sending the invite to the NP: remove_hf("Contact"); append_hf("Contact :<sip:XXX@server_ip :port;param1=p1;param2=p2>\r\n","Contact"); (The invite is sent perfectly and works great) After the NP reply i do: remove_hf("Contact"); append_hf("Contact :$ct","Contact"); Now I notice there are two contact headers. It seems that the remove_hf didn’t work this time. Any ideas? Thanks, Uri

2 1

Passwords in Kamailio user table are not encrypted.
by Albert Petit 27 Mar '12

27 Mar '12

good morning When provisioning subscriebrs to Kamailio with the kamctl tool i see the passwords are stored in clear on the database. For my use-case I would not like to have that passwords in clear on the db. What can I do to avoid this problem? Should I provision the users in a different way and not using kamctl tool or configure kamailio differently? If there is no way to achieve this would be dangerous if I edit the kamctl tool so it does not store any value on the password column and just store HA and HA1B values? BR,

2 1

Re: [SR-Users] Kamailio Load Testing
by Uri Shacked 26 Mar '12

26 Mar '12

> > > >How do you reload your htable? Is it using the exec( ) function to call > the > >mi command sht_reload? Or do you have another method? > > I use the exec() to run a shell script with the relevant mi command. the > shell script return code is used to make sure all went OK. > > >Thanks > > >Reda > > > > On Mon, Mar 26, 2012 at 14:56, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > > >* Hi, > *>* > *>* > *>* > *>* I made these kinds of tests before. I have two tips for you to pay > *>* attention to: > *>* > *>* 1. Read about the [routes] on SIPp, It is tricky to satisfy > kamailio > *>* with SIPp scenarios. > *>* > *>* 2. Make sure your DB table is well build (use less varchars and > more > *>* integers). > *>* > *>* I found out that there are many kamailio modules I can use to load > data in > *>* to memory and I hardly use the DB (only for ACC). I use MTREE, HTABLE, > *>* DIALPLAN, CARRIERROUTE, DROUTING, and so to store the information I > need. > *>* And I get it with the simple functions of the module. > *>* > *>* I load around 6 million numbers and other data which takes about 1.5 > Gb of > *>* the memory. > *>* > *>* For getting around the real-time changes that I need to deal with (In > DB > *>* the data changes are immediately made on your service…). I have a > RTIMER > *>* every 120 sec that check is a reload is needed. If so, it reloads the > *>* relevant data (very fast). > *>* > *>* BR, > *>* > *>* Uri > *>* > *>* > *>* > *>* > *>* Hello, > *>* > *>* On 3/20/12 12:12 PM, Stephen Dodge (Bistech) wrote: > *>* >* > *>* *>* Hello, > *>* *>* > *>* *>* I am running Kamailio 3.1.5 with a MySQL backend on CentOS. A > *>* *>* connection to MySQL (an off box MySQL cluster) is required for > every > *>* *>* call, sqlops is used to determine destinations and acc to record > CDR * > *>* >* information. > *>* *>* > *>* *>* I am planning to load test our server using SIPp to generate > calls, I > *>* *>* was wondering if anyone has done something similar and could > provide > *>* *>* advice on what we should monitor on our Kamailio Server. i.e > Server > *>* *>* CPU & Memory > *>* *>* > *>* *>* Thanks in advance for your help. > *>* *>* > *>* *if you do a lot of direct DB interactions, perhaps latency of the > *>* queries will be relevant. You can use benchmark module to see how long > *>* it takes to execute part of the config file. > *>* > *>* I think you don't fetch lot of records from db in config, so memory > *>* should be no problem, however you can dump private/shared memory usage > *>* via RPC commands within sercmd cli. CPU is a good metric always and > easy > *>* to watch with systems tools. > *>* > *>* Cheers, > *>* Daniel > *>* >* > *>* *>* Steve. > *>* * > * >

1 0

TLS delay on request processing
by Andreas Heise 26 Mar '12

26 Mar '12

Hi list, I use Kamailio 3.2.2 as TLS<=>TCP gateway this is working well, but if I create some traffic by SIPP then Kamailio response time increase with every new call, seems to be that Kamailio process call by call. I made a test with 30caps then after ~200calls Kamailio need ~3s to response an initial INVITE from the PBX. This delay seems to be increased with each call. test scenario: SIPP UAC---TCP---PABX---TLS---Kamailio---TCP---SIPP UAS 1.1.1.1 = PABX 2.2.2.2 = SIPP UAS 5.5.5.5 = Kamailio if ($si == "1.1.1.1") # from PABX { xlog("L_INFO","FROM PABX: $rm, $ci\n"); t_on_reply("1"); force_send_socket(tcp:5.5.5.5:5060); rewritehostport("2.2.2.2:5070"); t_relay_to_tcp("2.2.2.2","5070"); exit; }; I got all the calls in a single TLS connection from the PBAX so increasing of TCP childs doesn't help. I also already tried to adjust some TCP buffers and played with TCP_ASYNC. With UDP / TCP it's better, is really the TLS/openssl is the bottleneck or the way I relay the request? The test machine has only 512MB RAM and is a single core cpu (2,8GHZ), but 30caps isn't much? Any suggestions? many thanks, Andreas

2 2

Kamailio modifying ruri during loose-route
by Dan-Cristian Bogos 26 Mar '12

26 Mar '12

Hey Guys, I have noticed some unexpected behavior (at least by me) during my tests. When routing specific ACK message back to 200 OK originator, Kamailio will rewrite the ruri with the value of route header. Is that known or some memory access problem? Call setup: B2BUA(127.0.0.1) -> Kamailio (1.2.3.4) -> End device (2.3.4.5). Bellow is a trace of 200 OK -ACK flow before and after kamailio, together with the xlog capture of ruri before and after calling loose_route on ACK. Thanks in advance for any kind of tip. DanB PS: Since testing with a newer version would be a bit harder due to some particular features I use in 3.1 which are taking a bit of effort to migrate, have tried my luck here before, so hope to get your understanding. Kamailio version: """ root@kam:/home/dan# kamailio -V version: kamailio 3.1.5 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 07:41:01 Sep 15 2011 with gcc 4.4.5 """ Syslog: """ Mar 26 14:27:01 kam /usr/sbin/kamailio[7929]: ERROR: <script>: Before loose_route, ru is sip:01234567@127.0.0.1:5069;transport=udp Mar 26 14:27:01 kam /usr/sbin/kamailio[7929]: ERROR: <script>: Loose route, ru is sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes """ Ngrep of call: """ # U 2012/03/26 14:27:00.089455 127.0.0.1:5069 -> 1.2.3.4:25060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.2.3.4:25060;branch=z9hG4bK361e.ba530345.0. Via: SIP/2.0/UDP 192.168.1.29:5060;rport=3081;received=2.3.4.5;branch=z9hG4bK-373632-e2d6ebcc5a1d2a481e4957541b393f9a. Record-Route: <sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes>. From: "ines.demo" <sip:ines.demo@mydomain.net>;tag=a9c837bc. To: <sip:01234567@mydomain.net>;tag=BcBSHa56Bp7ge. Call-ID: 65464bffda1b33cfcc9d97c9e8f81cb3@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. Contact: <sip:01234567@127.0.0.1:5069;transport=udp>. User-Agent: WebiB2BUA 1.0.6. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 417. Remote-Party-ID: "01234567" <sip:01234567@mydomain.net>;party=calling;privacy=off;screen=no. . v=0. o=root 13076 13076 IN IP4 1.2.3.4. s=session. c=IN IP4 1.2.3.4. t=0 0. m=audio 48548 RTP/AVP 8 0 102 3 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 iLBC/8000. a=fmtp:102 mode=30. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=nortpproxy:yes. m=video 0 RTP/AVP 96 97 98. a=rtpmap:96 /0. a=rtpmap:97 /0. a=rtpmap:98 /0. # U 2012/03/26 14:27:00.089589 1.2.3.4:25060 -> 2.3.4.5:3081 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.29:5060;rport=3081;received=2.3.4.5;branch=z9hG4bK-373632-e2d6ebcc5a1d2a481e4957541b393f9a. Record-Route: <sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes>. From: "ines.demo" <sip:ines.demo@mydomain.net>;tag=a9c837bc. To: <sip:01234567@mydomain.net>;tag=BcBSHa56Bp7ge. Call-ID: 65464bffda1b33cfcc9d97c9e8f81cb3@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. Contact: <sip:01234567@127.0.0.1:5069;transport=udp>. User-Agent: WebiB2BUA 1.0.6. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 417. Remote-Party-ID: "01234567" <sip:01234567@mydomain.net>;party=calling;privacy=off;screen=no. . v=0. o=root 13076 13076 IN IP4 1.2.3.4. s=session. c=IN IP4 1.2.3.4. t=0 0. m=audio 48548 RTP/AVP 8 0 102 3 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 iLBC/8000. a=fmtp:102 mode=30. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=nortpproxy:yes. m=video 0 RTP/AVP 96 97 98. a=rtpmap:96 /0. a=rtpmap:97 /0. a=rtpmap:98 /0. # U 2012/03/26 14:27:00.160046 2.3.4.5:3081 -> 1.2.3.4:25060 ACK sip:01234567@127.0.0.1:5069;transport=udp SIP/2.0. Call-ID: 65464bffda1b33cfcc9d97c9e8f81cb3@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 192.168.1.29:5060;branch=z9hG4bK-373632-6eac5ecf0ba5ab4ce7094ad17870dca3. From: "ines.demo" <sip:ines.demo@mydomain.net>;tag=a9c837bc. To: <sip:01234567@mydomain.net>;tag=BcBSHa56Bp7ge. Max-Forwards: 70. Route: <sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes>. Contact: "ines.demo" <sip:ines.demo@192.168.1.29:5060;transport=udp;registering_acc=mydomain_net>. User-Agent: Jitsi1.0-beta1-nightly.build.3820Windows 7. Content-Length: 0. . # U 2012/03/26 14:27:00.160278 1.2.3.4:25060 -> 1.2.3.4:25060 ACK sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes SIP/2.0. Call-ID: 65464bffda1b33cfcc9d97c9e8f81cb3@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 1.2.3.4:25060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 192.168.1.29:5060;rport=3081;received=2.3.4.5;branch=z9hG4bK-373632-6eac5ecf0ba5ab4ce7094ad17870dca3. From: "ines.demo" <sip:ines.demo@mydomain.net>;tag=a9c837bc. To: <sip:01234567@mydomain.net>;tag=BcBSHa56Bp7ge. Max-Forwards: 69. Contact: "ines.demo" <sip:ines.demo@2.3.4.5:3081;transport=udp;registering_acc=mydomain_net>. User-Agent: Jitsi1.0-beta1-nightly.build.3820Windows 7. Content-Length: 0. """

2 1

Kamailio Load Testing
by Uri Shacked 26 Mar '12

26 Mar '12

Hi, I made these kinds of tests before. I have two tips for you to pay attention to: 1. Read about the [routes] on SIPp, It is tricky to satisfy kamailio with SIPp scenarios. 2. Make sure your DB table is well build (use less varchars and more integers). I found out that there are many kamailio modules I can use to load data in to memory and I hardly use the DB (only for ACC). I use MTREE, HTABLE, DIALPLAN, CARRIERROUTE, DROUTING, and so to store the information I need. And I get it with the simple functions of the module. I load around 6 million numbers and other data which takes about 1.5 Gb of the memory. For getting around the real-time changes that I need to deal with (In DB the data changes are immediately made on your service…). I have a RTIMER every 120 sec that check is a reload is needed. If so, it reloads the relevant data (very fast). BR, Uri Hello, On 3/20/12 12:12 PM, Stephen Dodge (Bistech) wrote: >* *>* Hello, *>* *>* I am running Kamailio 3.1.5 with a MySQL backend on CentOS. A *>* connection to MySQL (an off box MySQL cluster) is required for every *>* call, sqlops is used to determine destinations and acc to record CDR *>* information. *>* *>* I am planning to load test our server using SIPp to generate calls, I *>* was wondering if anyone has done something similar and could provide *>* advice on what we should monitor on our Kamailio Server. i.e Server *>* CPU & Memory *>* *>* Thanks in advance for your help. *>* *if you do a lot of direct DB interactions, perhaps latency of the queries will be relevant. You can use benchmark module to see how long it takes to execute part of the config file. I think you don't fetch lot of records from db in config, so memory should be no problem, however you can dump private/shared memory usage via RPC commands within sercmd cli. CPU is a good metric always and easy to watch with systems tools. Cheers, Daniel >* *>* Steve. *

2 1

2 rollover contacts per user
by Krishna Kurapati 26 Mar '12

26 Mar '12

Hi, Is there a way to set up maximum of 2 active roll over contacts per user? Currently when I setup max contacts param as 2, The third one is responding with 503 error for 3rd Register. How can we remove one of the contacts and replace it with the new one. I need it when the device is switching over from Wifi to 3G and back. or Wifi network to Wifi network switching. Thanks Krish Kura

3 5

Kamailio IMS + TLS
by trojak＠st.fri.uniza.sk 26 Mar '12

26 Mar '12

Hello, I am trying to add TLS support to kamailio PCSCF sever. After defining WITH_TLS value and editing tls.cfg file for right path to certificate and key I am still getting following errors: Mar 26 10:23:03 pcscf /usr/sbin/kamailio[12916]: ERROR: pcscf [mod.c:951]: ERR:P-CSCF:mod_init: get_tls_session_hash not found ! Mar 26 10:23:03 pcscf /usr/sbin/kamailio[12916]: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module pcscf (/usr/lib/kamailio/modules/pcscf.so) Mar 26 10:23:03 pcscf kamailio: ERROR: <core> [daemonize.c:269]: Main process exited before writing to pipe Is tls supported yet ? Will by packages in your repository updated to work with kamailio 3.2 ? Or any future updates of kamailio-ims-modules ? Thank you for help Best regards Lubo

2 1

question about cdrtool + freeradius
by Rabary 26 Mar '12

26 Mar '12

Hi all, I don't know if I'm in the right mailing, if not sorry in advance, if yes why I can't insert record in the mysql database for call accounting because it seems like the request is not complete. I put an attachment file, mysql log and /etc/freeradius/sql.conf. Thanks. -- Rabary Teddy Gulfsat Madagascar Service Technique +261-203-085-085 +261-345-600-080 Inutile d'imprimer ce mail

1 1

Absorbing ack's
by Egbert Groot 26 Mar '12

26 Mar '12

Hi all, I'm having an issue with ACK's being proxied. When an INVITE first comes in (new transaction, new dialog), I do a 'www_challenge(...)'. This makes the proxy send a '401 Unauthorized' reply. The client acknowledges this reply with "ACK". This ACK is 'absorbed' by the proxy. When during an established call/dialog the client sends an INVITE to set the call on hold, this is an in dialog INVITE (with to-tag etc). Although authentication for this INVITE isn't very necessary, my script does 'www_challenge' anyway. The proxy sends '401 Unauthorized', the client responds with 'ACK'. This ACK however is not 'absorbed' as it should be, but forwarded/proxied (or at least handed over to the routing script). Is this intended behaviour? If so, how can I recognize this ACK to be able to drop it? I found some related info in this email thread: http://lists.kamailio.org/pipermail/devel/2008-December/017246.html but doesn't discuss this exact issue. Though this isn't a big problem for me (the upstream gateway ignores the ack, and I can disable authentication for re-invite's), I'm curious why the proxy behaves this way. Hopefully someone can enlighten me. regards, Egbert.

2 3

Swetha Raj wants to chat
by Swetha Raj 25 Mar '12

25 Mar '12

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1 0

Kamailio Load Testing
by Stephen Dodge (Bistech) 24 Mar '12

24 Mar '12

Hello, I am running Kamailio 3.1.5 with a MySQL backend on CentOS. A connection to MySQL (an off box MySQL cluster) is required for every call, sqlops is used to determine destinations and acc to record CDR information. I am planning to load test our server using SIPp to generate calls, I was wondering if anyone has done something similar and could provide advice on what we should monitor on our Kamailio Server. i.e Server CPU & Memory Thanks in advance for your help. Steve. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

3 2

Re: [SR-Users] Accessing #!define "variables" from within lua
by Alex Balashov 24 Mar '12

24 Mar '12

I am not sure that exporting #defines to third-party language API modules makes much sense just because some user thought it would be neat. The #!... constructs are all intended as preprocessor directives, to make easier management of global constant values in the native Kamailio config environment and to allow easier management of conditionally-enabled chunks of code therein, right? Conceptually, it does not seem to me that they should have any representation outside of that evaluative context. It would be like making preprocessor macro values in C available through the libdl/ELF dynamic shared object interface. It doesn't make any sense. If you really want to export preprocessor macro-dependent values, trouble yourself to use an intermediate variable--a real variable. In Kamailio, this would take the form: #!ifdef MYPROJECT #!define MYPROJECT_IP 1.2.3.4 # Exportable symbol: myproject.ip = "MYPROJECT_IP" descr "..." #!endif Doesn't that make much more sense, in terms of what these directives were really intended for? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >Hello, > >On 3/23/12 2:00 PM, Asgaroth wrote: >> I'll check this option out, thanks for the tip Alex. >it is a good way to do, $sel(...) are available in Lua, @... format not, >but they are the same, just different access mechanism. > >> >> I guess the other option too, which I have not tried, is to possibly >> pass in the values as lua script parameters. > >Yes, you can do that. You may need to use 'subst' instead of 'define' in >order to be able to replace the token inside another string. > >Exporting defines to Lua could be a good enhancement anyhow, I will >think about and maybe add it in the near future. > >Cheers, >Daniel > >> >> On 23/03/12 12:12, Alex Balashov wrote: >>> Have you considered using selectors/cfg_get variables instead? >>> >>> They are also a common way to store various types of "global" >>> values/constants, with the difference that they can be accessed at >>> run time using PV container functions instead of set at preprocessing. >>> >>> myproject.local_ip = "1.2.3.4" descr "IP address" >>> >>> $sel(cfg_get.my project.local_ip) >>> Or SER-style: @cfg_get.my project.local_ip >>> >>> You can evaluate those in Lua, right? >>> >>> -- >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 235 E Ponce de Leon Ave >>> Suite 106 >>> Atlanta, GA 30030 >>> Tel: +1-678-954-0671 >>> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >>> >>> Asgaroth<00asgaroth00(a)gmail.com> wrote: >>> >>>> Hi All, >>>> >>>> I am looking at writing a lua function that would really benefit from >>>> accessing "variables" that I would have defined by using the #!define >>>> statement. Is this at all possible? I've tried looking at the >>>> documentation surrounding lua and kamailio, but I cannot see how I >>>> would >>>> access it. I thought I'd ask the list to see if anyone else may have >>>> attempted this. >>>> >>>> for example, I may have >>>> >>>> #!define INTERNAL_ADDRESS 1.2.3.4 >>>> >>>> Which I subsequently use through the the routing script to refer to the >>>> internal address I am listening on. Now I would like to try and access >>>> this "variable" from a lua script. Any pointers/tips/workarounds would >>>> be greatly appreciated. >>>> >>>> Thanks >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >-- >Daniel-Constantin Mierla >Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >http://www.asipto.com/index.php/kamailio-advanced-training/ > > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

www_authenticate("127.0.0.1", "credentials")
by Marwan Idriss 24 Mar '12

24 Mar '12

Greeting, I want to ask you how the function www_authenticate("127.0.0.1", "credentials") work and what should I put the credentials Table Regards Marwan

2 1

Re: [SR-Users] Accessing #!define "variables" from within lua
by Alex Balashov 24 Mar '12

24 Mar '12

Have you considered using selectors/cfg_get variables instead? They are also a common way to store various types of "global" values/constants, with the difference that they can be accessed at run time using PV container functions instead of set at preprocessing. myproject.local_ip = "1.2.3.4" descr "IP address" $sel(cfg_get.my project.local_ip) Or SER-style: @cfg_get.my project.local_ip You can evaluate those in Lua, right? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Asgaroth <00asgaroth00(a)gmail.com> wrote: >Hi All, > >I am looking at writing a lua function that would really benefit from >accessing "variables" that I would have defined by using the #!define >statement. Is this at all possible? I've tried looking at the >documentation surrounding lua and kamailio, but I cannot see how I would >access it. I thought I'd ask the list to see if anyone else may have >attempted this. > >for example, I may have > >#!define INTERNAL_ADDRESS 1.2.3.4 > >Which I subsequently use through the the routing script to refer to the >internal address I am listening on. Now I would like to try and access >this "variable" from a lua script. Any pointers/tips/workarounds would >be greatly appreciated. > >Thanks > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 2

Forward UAC Registration with Authentication (like Upper Registration or Passthrough Registration)
by Robert 24 Mar '12

24 Mar '12

Sorry to spam all 3 boards at the same time I know there are smart people on these lists and hoping that one might be willing to send an example config to assist a n00b to SER. I need to configure a proxy (like SBC) to listen on port X where X is not 5060. I want the UAC's registration to be forwarded (NOT REDIRECT) to the real SIP server on port 5060 The far end is not an Asterisk server so any examples of Asterisk do not work. It is not IP auth the authentication relies in the far end. Open to any solution of open source software K/OpenSIPS/OpenSBC etc. IP phone ----> open source -----> SIP server (UAC) ( PORT != 5060) ( PORT = 5060) I have tried all sorts of methods like rewritehostport / t_relay etc etc. Would uac_auth work? And if so anyone have any good examples of dynamic as I do not want to store any credentials in the open source proxy.

2 1

Re: [SR-Users] modify FROM field in the "failure_route" block
by Alex Balashov 24 Mar '12

24 Mar '12

I am going to have to go with the "upgrade" crowd in this one. Running OpenSER 1.3 at this point is like running a libc5-based Linux distro. Almost no advice or discussion is relevant to it because there have been such tectonic, generational shifts since. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Edson - Lists <4lists(a)gmail.com> wrote: >Please upgrade to latest stable and try again.... > >Why? If you have to ask it, than you didn't got it... > >new version === new functionalities > >99% of list support and tips are based on last stable version. 1.3 is >pretty old... > >Edson. > >Em 23/03/2012 10:06, Robert R escreveu: >> Hi, >> >> Even though when I try to use "uac_replace_from( )" in branch_route >> block I get the same error message: >> >> branch_route[2] >> { >> if ( $avp(i:204) ) >> { >> uac_replace_from("hello"); >> } >> >> } >> >> >> error message: >> >> Not starting openser: invalid configuration file! >> Mar 23 09:03:00 [7542] CRITICAL:core:yyerror: parse error in config >> file, line 623, column 26-27: Command cannot be used in the block >> Mar 23 09:03:00 [7542] ERROR:core:main: bad config file (1 errors) >> >> PS: I am running an old release of openser 1.3. >> >> Thanks, >> R >> >> >> >> >> On Thu, Mar 22, 2012 at 1:23 PM, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> >> >> On 3/22/12 6:12 PM, Alex Balashov wrote: >> >> On 03/22/2012 01:00 PM, Robert R wrote: >> >> When I am trying to modify FROM field in the "failure_route" >> block I >> get an error message: >> >> failure_route[1] >> { >> ... >> uac_replace_from("$avp(i:204)"__); >> append_branch(); >> t_relay(); >> } >> >> Not starting openser: invalid configuration file! >> Mar 22 10:51:39 [6457] CRITICAL:core:yyerror: parse error in >> config >> file, line 662, column 32-33: Command cannot be used in the >> block >> Mar 22 10:51:39 [6457] ERROR:core:main: bad config file (1 >> errors) >> >> >> You can probably get around it this way: >> >> failure_route[1] >> { >> ... >> >> append_branch(); >> >> route(STUFF); >> t_relay(); >> } >> >> >> route[STUFF] >> { >> uac_replace_from(...); >> } >> >> indeed this should work -- it is no reason of not being usable in >> failure_route, perhaps is so from the time of development. The >> function itself is allowed in branch_route which is called from >> inside t_relay() (when t_on_branch(...) is used), therefore we can >> relax the restriction for failure_route, imo. >> >> One thing to take care of, if you use uac_replace_from() in request >> route and then in failure route, you may end in malformed format, so >> it is recommended to use it in branch_route, then the changes done >> by the function will be per each forwarded branch. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >> http://www.asipto.com/index.__php/kamailio-advanced-__training/ >> <http://www.asipto.com/index.php/kamailio-advanced-training/> >> >> >> >> _________________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users >> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 2

Re: [SR-Users] Authentication on loose_route();
by Alex Balashov 24 Mar '12

24 Mar '12

Clearly, you can only authenticate sequential requests corresponding to calls whose initial requests were subject to authentication. If the initial request was not authenticated, there is no reason to believe that the endpoint would support authentication of sequential requests. As to whether you should do this, that is a controversial matter. I suppose that the security-maximising approach would be to challenge all requests, but it invites problems with many endpoints. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com David <kamailio.org(a)spam.lublink.net> wrote: > > >Hello, > >Should I be requiring users to authenticate before letting >them into loose_route(); ? What about anonymous calls from E164, how do >I authenticate these calls after they have started? > >Thanks, > >David > > > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Authentication on loose_route();
by David 23 Mar '12

23 Mar '12

Hello, Should I be requiring users to authenticate before letting them into loose_route(); ? What about anonymous calls from E164, how do I authenticate these calls after they have started? Thanks, David

1 0

cached $TV is 0 in failure route
by Klaus Darilion 23 Mar '12

23 Mar '12

Hi! Is this by design? I would expect it to be either the same value as in the request route, or even better to be calculated new every time it enters the failure route. regards Klaus

2 7

memory allocation failure while reading ca_list
by Juha Heinanen 23 Mar '12

23 Mar '12

in tls.cfg, i made ca_list to point to ca-certificates.crt file that comes with debian. when i then started sip router, i got errors Mar 19 08:45:42 test /usr/sbin/sip-proxy[28093]: ERROR: tls [tls_domain.c:447]: TLSc<default>: Unable to load CA list '/etc/ssl/certs/ca-certificates.crt' Mar 19 08:45:42 test /usr/sbin/sip-proxy[28093]: ERROR: tls [tls_domain.c:448]: load_ca_list:error:07064041:memory buffer routines:BUF_MEM_grow:malloc failure Mar 19 08:45:42 test /usr/sbin/sip-proxy[28093]: ERROR: tls [tls_domain.c:448]: load_ca_list:error:0906D041:PEM routines:PEM_read_bio:malloc failure Mar 19 08:45:42 test /usr/sbin/sip-proxy[28093]: ERROR: tls [tls_domain.c:448]: load_ca_list:error:0B084009:x509 certificate routines:X509_load_cert_crl_file:PEM lib Mar 19 08:45:42 test /usr/sbin/sip-proxy[28093]: ERROR: <core> [sr_module.c:889]: init_mod_child(): Error while initializing module tls (/usr/lib/sip-proxy/modules/tls.so) Mar 19 08:45:42 test /usr/sbin/sip-proxy[28093]: ERROR: <core> [main.c:1554]: ERROR: main: error in init_child(PROC_INT) -- exiting is that because there are too many (= 139) entries in ca-certificates.crt? if so, how to solve the problem? would it be a good idea that sip router would be default be able to handle this correctly? -- juha

3 14

modify FROM field in the "failure_route" block
by Robert R 23 Mar '12

23 Mar '12

Hi, When I am trying to modify FROM field in the "failure_route" block I get an error message: failure_route[1] { ... uac_replace_from("$avp(i:204)"); append_branch(); t_relay(); } Not starting openser: invalid configuration file! Mar 22 10:51:39 [6457] CRITICAL:core:yyerror: parse error in config file, line 662, column 32-33: Command cannot be used in the block Mar 22 10:51:39 [6457] ERROR:core:main: bad config file (1 errors) It seems "uac_replace_from()" can not be used in the "failure_route" block. What's the correct way to modify (appending a tag) FROM field in the "failure_route" block? Thanks, R

5 5

Search all dialogs - Kamailiio 3.0.3
by Charles Chance 23 Mar '12

23 Mar '12

Hi all, I was wondering if anyone could offer any advice please. We are using the dialog module and need to be able to find out whether another dialog exists within a particular profile and having a specific value, so that we can send the new call to the same gateway. As far as I am aware, the only way to see other dialogs from the request route is to query the database (having set db mode to realtime). This is ok, but the profile/value is not stored in the database table. Is there a way of fetching the information that I am missing, or should we just add some customisation to the module so that it also writes this information to db? Many thanks in advance, Charles

1 0

Accessing #!define "variables" from within lua
by Asgaroth 23 Mar '12

23 Mar '12

Hi All, I am looking at writing a lua function that would really benefit from accessing "variables" that I would have defined by using the #!define statement. Is this at all possible? I've tried looking at the documentation surrounding lua and kamailio, but I cannot see how I would access it. I thought I'd ask the list to see if anyone else may have attempted this. for example, I may have #!define INTERNAL_ADDRESS 1.2.3.4 Which I subsequently use through the the routing script to refer to the internal address I am listening on. Now I would like to try and access this "variable" from a lua script. Any pointers/tips/workarounds would be greatly appreciated. Thanks

1 0

Relaying message to Recent contact list...
by Krishna Kurapati 23 Mar '12

23 Mar '12

Hi, I am trying to relay the message to the Registered user. However, I would like to relay only to a recent contact. In case, there are multiple contact, I do not want to relay to older contacts. I am using t_relay() which is currently forking the request to all contacts. Any other way to accomplish my need? Thanks Krish Kura

2 3

Thank you from Google!
by resume-thanks＠google.com 23 Mar '12

23 Mar '12

1 0

ACCDB insert issues
by Uri Shacked 22 Mar '12

22 Mar '12

Hi, I am using the ACCDB with extra data. I collect the data to AVP's and set it on the db_extra. It gets to be 37 columns…. Pretty long… The problem I experience is that after a while (2-3 hours) with about 15 CAPS the MySQL ACC insert error appears. And, I am loosing accounting data. Except of narrowing the amount of columns and setting the fields (INT,CHAR etc.), does anyone has any ideas? The db_insert_mode set to 1 does not work (my ACC table is INNODB because I don’t want to lose info). BR, Uri

2 6

error when using "uac_replace_from( )" in the "failure_route" block
by Robert R 22 Mar '12

22 Mar '12

Hi, When I am trying to modify FROM field in the "failure_route" block I get an error message: failure_route[1] { ... uac_replace_from("$avp(i:204)"); append_branch(); t_relay(); } Not starting openser: invalid configuration file! Mar 22 10:51:39 [6457] CRITICAL:core:yyerror: parse error in config file, line 662, column 32-33: Command cannot be used in the block Mar 22 10:51:39 [6457] ERROR:core:main: bad config file (1 errors) It seems "uac_replace_from()" can not be used in the "failure_route" block. What's the correct way to modify (appending a tag) FROM field in the "failure_route" block? Thanks, R

3 2

Proxy Registrations offset port
by Robert 22 Mar '12

22 Mar '12

I've been playing with OpenSIPS and am willing to change over if it means I'll get a more informed answer from this community.. I need to set up a proxy forwarding server that can listen on a port other than 5060 I.e. 8080 and forward the registration requests to another Voip platform we have that is running on 5060. I've been able to do this under OpenSIPS but I'm having issues with inbound calls. It could be a NAT issue. Basically listen on 8080 forward any and all registrations to x:5060 which I've done using rewritehostport Anyone have any good example configs for this? I'm a n00b at this OpenSER stuff but have been around asterisk / freeswitch / portaone for a while. Thanks in advance

6 12

Does anyone know how to use Sipp to test Kamailio?
by Jiaxiang Zhou 22 Mar '12

22 Mar '12

Hi I want to do the performance testing on the Kamailio. I found the related document about openser 1.2.0 version at: http://www.kamailio.org/docs/openser-performance-tests/. But somehow, I can't make it work with Kamailio 3.1. Do you happen to know what to do? By following the test case at http://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c, I use the command as below to do the test: ./sipp -sf uac_inv.xml<http://www.kamailio.org/docs/openser-performance-tests/data/uac_inv.xml>-rsa 10.1.11.24:5060 192.10.1.11.24:5081 -m 200000 -r 7000 -d 1 -l 27 where 10.1.11.24:5060 is for Kamailio, and 10.1.11.24:5081 is for sipp uas. In the error log, I got the error message: *********************************************************************'. 2012-03-22 08:55:43:933 1332377743.933861: Aborting call on unexpected message for Call-Id '77062-11425(a)172.16.220.1': while expecting '100' (index 1), received 'SIP/2.0 403 Not relaying Via: SIP/2.0/UDP 172.16.220.1:5060 ;branch=z9hG4bK-11425-77062-0;rport=5060;received=10.1.11.197 From: sipp <sip:sipp@172.16.220.1:5060>;tag=11425SIPpTag0077062 To: sut <sip:service@10.1.1.197:5080 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.489a Call-ID: 77062-11425(a)172.16.220.1 CSeq: 1 INVITE Server: kamailio (3.2.0 (x86_64/linux)) Content-Length: 0 '. 2012-03-22 08:55:43:933 1332377743.933965: Aborting call on unexpected message for Call-Id '77063-11425(a)172.16.220.1': while expecting '100' (index 1), received 'SIP/2.0 403 Not relaying Via: SIP/2.0/UDP 172.16.220.1:5060 ;branch=z9hG4bK-11425-77063-0;rport=5060;received=10.1.11.197 From: sipp <sip:sipp@172.16.220.1:5060>;tag=11425SIPpTag0077063 To: sut <sip:service@10.1.1.197:5080 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.853f Call-ID: 77063-11425(a)172.16.220.1 CSeq: 1 INVITE Server: kamailio (3.2.0 (x86_64/linux)) Content-Length: 0 ********************************************************************* Thanks, Josh

2 1

Re: [SR-Users] ACCDB insert issues
by Yufei Tao 22 Mar '12

22 Mar '12

Hi I've also seen this error "ERROR: acc [acc.c:405]: failed to insert into database ". This seem to happen when cdrs_procedure runs at the same time of acc table insert. Can also be reliably reproduced by locking the acc table, make a call (about 20 seconds or more), end the call, wait for a short while, unlock acc table. This causes this error, and Kamailio inserts duplicated INIVTE / BYE entries into acc table. The number of duplicated entries depends on how long you've locked the table for. So I think this acc insert error may have been to do with table locking when cdrs_procedure is run. When there are duplicated entries in acc table, cdrs_procedure takes the first INVITE/BYE entires and updated the cdrs table correctly. But threw an error when it hits the next duplicated entry and stop. So for us it didn't seem to cause any trouble for cdrs records so I didn't chase it further. But would be nice if we could stop/handle the duplicated entries some how. Not a fix at at all, just some clues that might help. Yufei Message: 5 Date: Wed, 21 Mar 2012 20:28:33 +0200 From: Uri Shacked <ushacked(a)gmail.com> Subject: Re: [SR-Users] ACCDB insert issues To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <CAMMbDhTK2h4s1y_YJ_Tp8J7eZT4cmn_w1OkDBAS+3oAaZEZpAw(a)mail.gmail.com> Content-Type: text/plain; charset="windows-1252" Well, after some experiments... Two things made the difference: 1. Use only the id as Pkey and remove all other indexes (helped a little). 2. Set myisam engine to the table (i knew that but tried innodb for its reliability) Now we are back in order to deal with many, many CAPS. Thanks, If someone has another idea, and uses innodb for the ACC. please advise... BR, Uri On Tue, Mar 20, 2012 at 12:22 PM, Uri Shacked <ushacked(a)gmail.com> wrote: > > no log on the msqld for it. > > there is the CDR server that collects the events every 5 min and deletes > > old once... i disabled it becoase i believed it was it. but, no.... the > > error apears again... > > a BI specialist told me it must have something to do with the number of > > columns and the data type... > > > > On Tue, Mar 20, 2012 at 12:06 PM, Daniel-Constantin Mierla < > > miconda(a)gmail.com> wrote: > > >> >> Do you have error messages in mysql server logs? Is there another >> >> application that may block the acc table? >> >> >> >> The error refers to unavailability of the mysql server... >> >> >> >> Cheers, >> >> Daniel >> >> >> >> >> >> On 3/20/12 11:03 AM, Uri Shacked wrote: >> >> >> >> ERROR: db_mysql [km_dbase.c:122]: driver error on query: Lost >> >> connection to MySQL server during query >> >> Mar 18 20:03:38 net-ivr-KamIN-J1 /usr/local/sbin/kamailio[2671]: ERROR: >> >> <core> [db_query.c:186]: error while submitting query >> >> Mar 18 20:03:38 net-ivr-KamIN-J1 /usr/local/sbin/kamailio[2671]: ERROR: >> >> acc [acc.c:405]: failed to insert into database >> >> >> >> Whe apears, from time to time, for some of them. it can rest for 2-10 >> >> hours, then have a "glitch" of 160 sql driver errors, and rest again.... >> >> >> >> >> >> >> >> On Tue, Mar 20, 2012 at 11:51 AM, Daniel-Constantin Mierla < >> >> miconda(a)gmail.com> wrote: >> >> >>> >>> Hello, >>> >>> >>> >>> >>> >>> On 3/20/12 10:11 AM, Uri Shacked wrote: >>> >>> >>> >>> Hi, >>> >>> >>> >>> >>> >>> >>> >>> I am using the ACCDB with extra data. I collect the data to AVP's and >>> >>> set it on the db_extra. It gets to be 37 columns?. Pretty long? >>> >>> The problem I experience is that after a while (2-3 hours) with about 15 >>> >>> CAPS the MySQL ACC insert error appears. >>> >>> And, I am loosing accounting data. >>> >>> >>> >>> what are the errors? Once they appear, is it for all records or just >>> >>> from time to time, to some of them? >>> >>> >>> >>> Cheers, >>> >>> Daniel >>> >>> >>> >>> >>> >>> >>> >>> Except of narrowing the amount of columns and setting the fields >>> >>> (INT,CHAR etc.), does anyone has any ideas? >>> >>> >>> >>> >>> >>> >>> >>> The db_insert_mode set to 1 does not work (my ACC table is INNODB >>> >>> because I don?t want to lose info). >>> >>> >>> >>> >>> >>> >>> >>> BR, >>> >>> >>> >>> Uri >>> >>> >>> >>> >>> >>> _______________________________________________ >>> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> >>> >>> -- >>> >>> Daniel-Constantin Mierla >>> >>> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germanyhttp://www.asipto.com/index.php/kamailio-advanced-training/ >>> >>> >>> >>> >> >> >> >> -- >> >> Daniel-Constantin Mierla >> >> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germanyhttp://www.asipto.com/index.php/kamailio-advanced-training/ >> >> >> >> > >

1 0

need help for newbie in Kamailio
by Rabary 21 Mar '12

21 Mar '12

Hi mailing, Newbie to kamailio, I follow this tuto http://nil.uniza.sk/sip/kamailio/adding-mysql-support-kamailio-31-debian-le… for the registration SIP via mysql database and it works fine, but I saw that when during the call we disconnect the called UAC from network or turn the power off the caller UAC don't hangup. Is there any tool for how to hangup call when the UAC on the other side has no network connection or it isn't power on durring a call ? I heard for mediaroxy or rtpproxy but I don't know if them can do what I except to haveand we also use ip routing to make kamailio server to communicate with the UAC so we don't use NAT. Our topology is: kamailio (with public IP address) ---> cisco switch ---> LAN ---> UAC (with private IP address) Thanks in advance. -- Inutile d'imprimer ce mail

5 9

Kamailio NP module
by Ahmed 21 Mar '12

21 Mar '12

Hi, I try to run this module, but unsuccesfully. Can you give me some advices? How I can start the pdb_server? I compile and install it, but don´t know how get it run. I must add loadmodule="pdb.so" in kamailio.cfg file? When I did it, kamailio didnt start. Starting end with error. I think that there missing bracket : # query external service for routing information if (!pdb_query("$rU", "$avp(i:82)")) $avp(i:82) = 0; # default routing } cr_route("$avp(i:82)", "$rd", "$rU", "$rU", "call_id"); Thank you for your answer. Have a nice day.

2 1

Pdb module
by Ahmed 20 Mar '12

20 Mar '12

Hi, I try to run this module, but unsuccesfully. Can you give me some advices? How I can start the pdb_server? I compile and install it, but donÂ´t know how get it run. I must add loadmodule="pdb.so" in kamailio.cfg file? When I did it, kamailio didnt start. Starting end with error. I think that there missing bracket : # query external service for routing information if (!pdb_query("$rU", "$avp(i:82)")) $avp(i:82) = 0; # default routing } cr_route("$avp(i:82)", "$rd", "$rU", "$rU", "call_id"); Thank you for your answer. Have a nice day.

1 0

Request-URI in fail route
by Jan Klepal 20 Mar '12

20 Mar '12

Hello, I am trying to set simple no-answer forward scenario: Alice calls Bob and after timeout 5s forward to Carol (and additionally Carol timeout forward to Dave). I am using Kamailio 3.2.2 default config. There is part of routing logic for no-answer forwarding: route[RELAY] { if(is_method("INVITE")) { t_set_fr(5000); t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if(!t_relay()) { sl_reply_error(); } } branch_route[BRANCH_ONE] { $avp(divert) = $null; if($rU == "bob") { $avp(divert) = "carol"; } # second problem if($rU == "carol") { $avp(divert) = "dave"; } } failure_route[FAIL_ONE] { if($avp(divert) != $null) { add_diversion("no-answer"); $rU = $avp(divert); route(LOOKUP); route(RELAY); } } First problem I encountered is that I can't check $rU in failure_route (it is always set to original Request-URI - bob). Is this normal behavior or am I missing something? Because of that I have to set divert destination in branch route. Second problem is when Carol's no-answer forward to Dave happens. Then add_diversion adds invalid diversion header because it uses Request-URI which is still set to bob. Kind Regards, Jan

2 5

server overload mitigation
by Vineet Menon 20 Mar '12

20 Mar '12

Hi, does kamailio mitigate server overload???If yes how is that?? Is there a setting for high water mark or something?? Regards, Vineet Menon

2 1

FW: Unsuccessful call using DRouting - Request Timeout
by Fatima Chahrour~Vanrise Support 20 Mar '12

20 Mar '12

Dears, I started my lab on VMware (debian lenny) where I installed Kamailio 3.2 and started drouting configuration. I'm using 2 PCs each having an x-lite softphone and the Kamailio on the vmware all connected locally. My primary test is to make a call from x-lite on PC (192.168.111.11) to x-lite on PC (192.168.111.226) through Kamailio\ drouting modules (192.168.111.139). What is happening is that when I call 9611558800, for example, am getting 'Calling' message on PC x-lite followed by 'Call Failed: Request Timeout' after a certain time. Attached is my routing logic script, and ngrep for the call that is failing. And my dr_tables data is : dr_rules: 4, '0', '9611', '', 0, '', '#3', 'Lebanon Fix' dr_gw_lists: 3, '5', 'Test Call' dr_groups: 3, '.*', '.*', 0, 'Everyone' dr_gateways: 5, 1, '192.168.111.226', 0, '', '', 2, 'PC1' I tried adding port number to the gateways, remove '#' and add 5 in the dr_rules, remove the "0" from do_routing in the script, but with no success. Reading some related documents and books, I found that all scenarios using drouting have gateways as media servers (Asterisks) . So my question: is this a must, to have a media proxy even using my above primary enviroment? If not, can you please help me finding my mistake that prohibit the call's success(ringing)? Thanks in advance. F.Chahrour

3 5

Dispatcher and round-robin ignoring priorities?
by Øyvind Kolbu 20 Mar '12

20 Mar '12

Our PSTN provider has three servers. Two of them are located in our town, and we prefer them, and the last one much further away. We can use them interchangeably. >From our dispatcher file: # 10: PSTN 10 sip:preferred1.example.com:7476 0 5 10 sip:preferred2.example.com:7476 0 5 10 sip:third.example.com:7476 0 4 In our dispatch router we use 'ds_select_dst("10","4")'. Then the round-robin algorithm evenly distributes the load on all three of them, instead of round robin on the ones with priority 5 and fallbacking to priority 4. Is the unconditional round robin as excepted? When changing the selection algorithm to "8", «use first destination», the dispatcher module correctly uses first "preferred2", then "preferred1" and then lastly "third". Why «first destination» is not the first one listed in the dispatcher file has always puzzled me.. # kamailio -V version: kamailio 3.2.1 (i386/linux) 31c991 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 31c991 compiled on 08:08:06 Dec 6 2011 with gcc 4.1.2 -- Øyvind Kolbu

2 2

Users being registered without any Authentication
by Faisal Rehman 20 Mar '12

20 Mar '12

Hi, I have the below configuration file kamailio.cfg but all the users are not being registered with authentication, so am I doing some mistake in it? #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://faisal:faisal123@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags #FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ /* 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:192.162.15.23:5161 listen=tcp:129.162.15.23:5161 alias="xmpp.kamailio.org" /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5161 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "xmpp.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) #--------XMPP Module Params--------------- modparam("xmpp", "domain_separator", "%") modparam("xmpp", "xmpp_domain", "192.168.15.23") modparam("xmpp", "xmpp_host", "192.168.15.23") modparam("xmpp", "gateway_domain", "192.168.15.23") modparam("xmpp", "backend", "server") modparam("xmpp", "xmpp_port", 5299) modparam("xmpp", "xmpp_password", "casilla233") # ----- 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) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* 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", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); #authentication route(AUTH); ###############-----------------------------------XMPP Server Configuration----------------------------------------################# if (uri =~ "sip:.+@sip-xmpp\.kamailio\.org") { #absorb transmission if (!t_newtran()){ sl_reply_error(); exit; } #Handling Instant Messaging if (is_method("MESSAGE")){ xlog("message to XMPP: from <$fu> ru>\n"); if (xmpp_send_message()){ sl_send_reply("202", "Accepted"); } else { sl_send_reply("404", "Not found"); } exit; } #Un-supported Type of SIP Message xlog("message to XMPP: method [$rm] not supported yet\n"); sl_send_reply("503", "Service unavailable"); exit; } ##################--------------------------------XMPP Server Configuration-----------------------------------------------############### # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # 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 route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } Regards, Faisal

3 5

Presence question
by David 19 Mar '12

19 Mar '12

Hello, I am running a presence server. I am searching for a problem with my presence, and I have traced it down to this query : delete from active_watchers where presentity_uri='sip:test109@domain.tld' AND event='dialog' AND to_tag='f0253efb8005057b3cc0ab8fa35ff1d9-c961' This query is being sent to the database a few seconds after the initial subscribe is received. So the subscription only lasts in the database for a few seconds. I tried running level 3 error reporting ( hundreds of DEBUG ). I did not see any justification for query. Where is this query coming from ( source code/ config file )? Why is this subscription being deleted almost immediatly after it is setup ? Thanks, David Here is a sample request ( not the same as the one above, I did not capture the SIP at the same time as the SQL ) : U PROXYIP:5060 -> PRESENCEIP:5060 SUBSCRIBE sip:test109@domain.tld SIP/2.0. Record-Route: <sip:PROXYIP;lr=on;ftag=db911ce615f1598c>. Record-Route: <sip:TCPPROXYIP;r2=on;lr=on>. Record-Route: <sip:TCPPROXYIP;transport=tcp;r2=on;lr=on>. Via: SIP/2.0/UDP PROXYIP;branch=z9hG4bK9543.f0668033.0. Via: SIP/2.0/UDP TCPPROXYIP;branch=z9hG4bK9543.d2e18073.0;i=0a3. Via: SIP/2.0/TCP USERIP:61334;branch=z9hG4bK-6ee64c28;rport=61334. To: sip:test109@domain.tld. From: "103 Zone T" <sip:test103@domain.tld>;tag=db911ce615f1598c. Call-ID: 3f55db6e-b2ec6ca4(a)192.168.219.109. CSeq: 53944 SUBSCRIBE. Max-Forwards: 68. Contact: "103 Zone T" <sip:test103@USERIP:8280;transport=tcp>. Accept: application/dialog-info+xml. Expires: 1800. Event: dialog. User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. . # U PRESENCEIP:5060 -> PROXYIP:5060 SIP/2.0 202 OK. Record-Route: <sip:PROXYIP;lr=on;ftag=db911ce615f1598c>. Record-Route: <sip:TCPPROXYIP;r2=on;lr=on>. Record-Route: <sip:TCPPROXYIP;transport=tcp;r2=on;lr=on>. Via: SIP/2.0/UDP PROXYIP;branch=z9hG4bK9543.f0668033.0. Via: SIP/2.0/UDP TCPPROXYIP;branch=z9hG4bK9543.d2e18073.0;i=0a3. Via: SIP/2.0/TCP USERIP:61334;branch=z9hG4bK-6ee64c28;rport=61334. To: sip:test109@domain.tld;tag=f0253efb8005057b3cc0ab8fa35ff1d9-8c10. From: "103 Zone T" <sip:test103@domain.tld>;tag=db911ce615f1598c. Call-ID: 3f55db6e-b2ec6ca4(a)192.168.219.109. CSeq: 53944 SUBSCRIBE. Expires: 1800. Contact: <sip:PRESENCEIP:5060>. Server: kamailio (3.1.5 (x86_64/linux)). Content-Length: 0.

2 1

Re: [SR-Users] XMPP module not passing subscribes coming from xmpp side
by Dan-Cristian Bogos 19 Mar '12

19 Mar '12

Hi Pavel, Many thanks for your reaction. Good to know about the issues, saved me some important time with troubleshooting. Will try to find alternatives in this case since I already found some other limitations which stop me in having an enterprise solution. Have a good one! DanB > > Hi Dan, > > student of mine (Martin) realized that, but it was using Jabberd2 not > opernfire XMPP server, because the Openfire did not register domain from the > XMPP component correctly and did some other strange things. > Here are parts of Martin communication > > http://comments.gmane.org/gmane.comp.voip.openser.user/31713 > > > palo73 > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Dan-Cristian Bogos >> Sent: Friday, March 02, 2012 11:22 AM >> To: sr-users(a)lists.sip-router.org >> Subject: [SR-Users] XMPP module not passing subscribes coming from xmpp >> side >> >> Hey Guys, >> >> I was wondering if anyone having experience with using xmpp module as >> presence gateway between SIP and XMPP. I started evaluating it and till >> now got some weird results, not sure if it is just my setup which does >> not match the one widely used or the module is incomplete. >> >> Regarding my setup, I use OpenFIRE 3.7.1 on one side and Kamailio 3.2.0 >> out of debian packages on the other. >> >> The issue I got right now is that the subscribe coming from XMPP side >> produces no action/reply on SIP side. Bellow you can see the packet >> coming from OpenFIRE: >> # >> T 2012/03/02 09:49:57.743750 127.0.0.1:5275 -> 127.0.0.1:49965 [AP] >> <presence id="4h8F6-19" to="dan4(a)gw.mydomain.com" type="subscribe" >> from="dan(a)mydomain.com"><c xmlns="http://jabber.org/protocol/caps" >> ext="voice-v1 video-v1 camera-v1 " hash="sha-1" >> node="http://jitsi.org" >> ver="ohjz8WKq0ZQRiNWRNcapX4BBHLk="/></presence> >> """ >> >> Anyone here got this working somehow? >> >> Ta, >> DanB >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi- >> bin/mailman/listinfo/sr-users > > > > > ------------------------------ > > Message: 5 > Date: Fri, 2 Mar 2012 11:44:12 +0100 > From: Reda Aouad <reda.aouad(a)gmail.com> > Subject: [SR-Users] Registration Limits > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAA30pc7pkPcBSfMSdRpvCSHNm0kiSw9r13huUe4jikiEPdcr1Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Is there a way to ensure single-registration per user-agent for a user, > which overwrites previous registration ? > Or is there a way to limit the number of registrations per user, but > overwriting the earliest registration for each new one ? > > Thanks, > Reda >

3 4

Unsuccessful call using DRouting - Request Timeout
by Fatima Chahrour~Vanrise Support 19 Mar '12

19 Mar '12

Dears, I started my lab on VMware (debian lenny) where I installed Kamailio 3.2 and started drouting configuration. I'm using 2 PCs each having an x-lite softphone and the Kamailio on the vmware all connected locally. My primary test is to make a call from x-lite on PC (192.168.111.11) to x-lite on PC (192.168.111.226) through Kamailio\ drouting modules (192.168.111.139). What is happening is that when I call 9611558800, for example, am getting 'Calling' message on PC x-lite followed by 'Call Failed: Request Timeout' after a certain time. Attached is my routing logic script, and ngrep for the call that is failing. And my dr_tables data is : dr_rules: 4, '0', '9611', '', 0, '', '#3', 'Lebanon Fix' dr_gw_lists: 3, '5', 'Test Call' dr_groups: 3, '.*', '.*', 0, 'Everyone' dr_gateways: 5, 1, '192.168.111.226', 0, '', '', 2, 'PC1' I tried adding port number to the gateways, remove '#' and add 5 in the dr_rules, remove the "0" from do_routing in the script, but with no success. Reading some related documents and books, I found that all scenarios using drouting have gateways as media servers (Asterisks) . So my question: is this a must, to have a media proxy even using my above primary enviroment? If not, can you please help me finding my mistake that prohibit the call's success(ringing)? Thanks in advance. F.Chahrour

1 0

message offline store questions...
by Krishna Kurapati 19 Mar '12

19 Mar '12

Hi, I am using msilo module to store offline message. I am sending back "202 pending" when the message is stored (m_store) However, when I message delivered (m_dump), how do I inform the sender that the message is successfully delivered? Also, how do I handle if the sender if offline when I try to inform about the delivery of the message. I think I need offline notify store to handle this condition. Instead of removing the message from the silo table upon receipt of 200 OK from the receiver, I am thinking keep the message there until the NOTIFY is successfully delivered to the sender. In effect, the silo table in the database function as offline message and notify store. I reviewed module_k/msilo source code, and the functionality that I need seems missing. I am thinking of two approaches. 1) Extend the msilo module to implement the feature, if more people need it, then I can submit the changes. 2) Write sqlops + lua script to address the missing function. In either case, I am thinking of storing the callid of the original message (possibly in silo table?) in the database and sending the NOTIFY with the callid to the sender. I would like to extend the silo table with two columns: 1. the callid of the message 2. flags to indicate status of the delivery of the message. Appreciate your comments, suggestions and alternatives. Thanks Krish Kura

2 2

Re: [SR-Users] DRouting cause Kamailio 3.2 to stop
by Fatima Chahrour~Vanrise Support 19 Mar '12

19 Mar '12

Dear Daniel, Thanks for your reply.. I'd like to ask you too if my script in my first post that am using in the routing area is correct? I found file named core in the '/' location as you mentioned. When I open it using #cat core, the opened file go crazy and writes strange characters and it ending with the below: /usr/local/sbin/kamailio-P/var/run/kamailio.pid-f/usr/local/etc/kamailio//ka mailio.cfgSSH_AGENT_PID=3234GPG_AGENT_INFO=/tmp/seahorse-jKaZEU/S.gpg-agent: 3246:1TERM=xtermDESKTOP_STARTUP_ID=SHELL=/bin/bashGTK_RC_FILES=/etc/gtk/gtkr c:/home/vanrise/.gtkrc-1.2-gnome2WINDOWID=33580848USER=rootGNOME_KEYRING_SOC KET=/tmp/keyring-ZKxFck/socketSSH_AUTH_SOCK=/tmp/keyring-ZKxFck/sshUSERNAME= vanriseSESSION_MANAGER=local/debian:/tmp/.ICE-unix/3188MAIL=/var/mail/rootPA TH=/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/DESKTOP_SES SION=defaultGDM_XSERVER_LOCATION=localPWD=/usr/local/etc/kamailioLANG=en_US. UTF-8GDM_LANG=en_US.UTF-8GDMSESSION=defaultSHLVL=4HOME=/rootGNOME_DESKTOP_SE SSION_ID=DefaultLOGNAME=rootDBUS_SESSION_BUS_ADDRESS=unix:abstract=/tmp/dbus -sH8atcWdX4,guid=f0dcba443f8fd1d5267161364f4dd99dXDG_DATA_DIRS=/usr/local/sh are/:/usr/share/:/usrdebian:/# 62;9;c62;9 Sorry, I've just realized an info that I should mention and would be helpful, am running this Kamailio on a debian Lenny VMware as a trial before migrating it to its server. Thank you, F.Chahrour From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, March 01, 2012 10:15 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Fatima Chahrour~Vanrise Support; sr-users-bounces(a)lists.sip-router.org Subject: Re: [SR-Users] DRouting cause Kamailio 3.2 to stop Hello, have you found the core file in /var/log/? It is a strange location unless you started with -w /var/log. The core file is named usually 'core' or core.pid (replace pid with same number, like, 'core.1234'). The default location in in '/' unless there is a different location set via -w or enforced by operating system (e.g., mac os x puts it in '/corefiles'). Hope it helps to locate the core file and send the backtrace. Cheers, Daniel On 3/1/12 9:05 AM, Fatima Chahrour~Vanrise Support wrote: I also tried : #gdb /usr/local/sbin/kamailio /var/log/core.file (gdb) bt And got No Stack.. BR, F.Chahrour

2 9

ser_mysql.sh
by Marwan Idriss 19 Mar '12

19 Mar '12

Greeting, I need the ser_mysql.sh for the ser version ser 2.0.0-4ubuntu2 Regards Marwan

1 0

Best approach for TCP/TLS connection re-use for nated Contacts
by Giacomo Vacca 19 Mar '12

19 Mar '12

Hi all, I have a scenario where it's not clear to me what the best approach should be (Kamailio 3.2.0 on Squeeze, although I think it's a matter of routing logic only). I have a client behind NAT that sends an initial REGISTER request with a Contact containing private ip:port (e.g. Contact: "GV" <sip:12345@192.168.1.2:54321;transport=tcp> ) However, after it's challenged for authentication with a 401, it sends the following REGISTER request with Authorization header and with a Contact containing ip:port equal to the 'received' SIP URI (with a combination of 'rport' and 'received' parameters (e.g. Contact: "GV" <sip:12345@[receivedIP]:[rport];transport=tcp> ) In this case kamailio's routing logic is not marking the REGISTER as natted (no Received field in location and no NAT flags set). You can imagine I have a fairly standard NAT handling: route[NATDETECT] { force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } return; } FLT_NATS won't be set, and so it won't FLB_NATB in the following processing. This client receives a call: Kamailio relays the INVITE to it using the stored Contact. The INVITE is correctly received by the client, because the stored Contact refers to the connected socket ([received]:rport). The trouble starts when then the client replies with a 200 OK containing the private ip:port in the Contact header field (again 192.168.1.2:54321). The ACK will then have that private ip:port in the R-URI, and its relaying will fail. One of the solutions I've found is using always add_contact_alias() in onreply_route when handling the 200 OK, and then use handle_ruri_alias() when defining the destination for the ACK. I know that without traces and the full .cfg this is quite a broad question, but I wonder if you have already had to deal with this kind of client behaviour and can provide advice on the best practices. This is in particular useful for the re-use of TCP-based connections. Thanks in advance, Giacomo Truphone Limited is a limited liability company registered in England & Wales, registered office: 4 Royal Mint Court, London, EC3N 4HJ. Registered No. 04187081. VAT No. GB 851 5278 19. Tru is a brand name of Truphone and is a Truphone Communications Service. Truphone is a trading name for a number of distinct legal entities that operate in combination. www.truphone.com<http://www.truphone.com>. This e-mail, and any attachment(s), may contain information which is confidential and/or privileged, and is intended for the addressee only. If you are not the intended recipient, you may not use, disclose, copy or distribute this information in any manner whatsoever. If you have received this e-mail in error, please contact the sender immediately and delete it.

3 3

register_udomain(): Invalid table version (use ser_mysql.sh reinstall)
by Ryan Gholam 19 Mar '12

19 Mar '12

Hello , I hope someone encountered this problem on UBUNTU , when connecting to the database i get this error in my log file :var/log/sys Mar 19 14:09:46 rayan-desktop /usr/sbin/ser[21807]: register_udomain(): Invalid table version (use ser_mysql.sh reinstall) Mar 19 14:09:46 rayan-desktop /usr/sbin/ser[21807]: domain_fixup(): Error while registering domain Any help is appreciated . Thank you .

1 0

add_contact_alias and handle_ruri_alias vs fix_nated_contact
by yufei.tao 19 Mar '12

19 Mar '12

Hi I have been using fix_nated_register/fix_nated_contact for NAT traversal and all worked fine. I've come across add_contact_alias and handle_ruri_alias: http://sip-router.org/wiki/tutorials/alias-example where it says: "The benefits of using add_contact_alias() and handle_ruri_alias() functions instead of conventional NAT traversal solutions are: * Request-URI in requests to UAs behind NATs is always what UAs expect * Re-use of tcp/tls sessions between proxy and UAs " I assumed the 'conventional NAT traversal solution' here means fix_nated_contact? I understand the first point, in that the R-URI always contains what the client puts in the contact hf. But for the second point, about tcp/tls reuse, does fix_nated_contact do the same trick, or does add_contact_alias and handle_ruri_alias give you more benefit over fix_nated_contact? So far I feel they achieve the same thing but would like this confirmed/corrected. Thank you very much! Yufei

4 3

Re: [SR-Users] dns lookup and store IP in avp..
by MingHon 19 Mar '12

19 Mar '12

On Mon, Mar 19, 2012 at 4:56 PM, MingHon <gminghon(a)gmail.com> wrote: > Thanks Daniel... :) > > > On Mon, Mar 19, 2012 at 4:18 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> >> On 3/19/12 8:22 AM, Klaus Darilion wrote: >> >>> No. If you really need it, then you can use exec() and a tool like "dig" >>> or "host". >>> >> apart of exec module functions, another option would be using an embedded >> language script (Lua/Perl...). >> >> Cheers, >> Daniel >> >>> >>> regards >>> Klaus >>> >>> On 19.03.2012 03:19, MingHon wrote: >>> >>>> Hi All, >>>> >>>> is this function available?? >>>> >>>> http://sourceforge.net/**tracker/?func=detail&aid=** >>>> 2687695&group_id=139143&atid=**743023<http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati…> >>>> <http://sourceforge.net/**tracker/?func=detail&aid=** >>>> 2687695&group_id=139143&atid=**743023<http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati…>> >>>> >>>> >>>> Thanks :) >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> MingHon >>>> >>> >>> ______________________________**_________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >> -- >> Daniel-Constantin Mierla >> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >> http://www.asipto.com/index.**php/kamailio-advanced-**training/<http://www.asipto.com/index.php/kamailio-advanced-training/> >> >> > > > -- > Regards, > > MingHon > -- Regards, MingHon

1 1

Memory Question
by Klaus Darilion 19 Mar '12

19 Mar '12

Hi! Started kamailio with "-m 100 -M 30". Thus I would suspect that every process needs 30MB, and one of the processes needs 30+100Mb as it allocates the shared memory. The further would conclude that the memory will be allocated only once during startup. Can this be somehow verified? Normal "ps" output shows that every process needs ~130MB virtual memory, but this is probably not what I am looking for. Thanks Klaus # ps aux PID %CPU %MEM VSZ RSS TTY STAT COMMAND 846 0.2 0.8 139732 4112 ? S kamailio -m 100 -M 30 847 0.0 0.2 139732 1188 ? S kamailio -m 100 -M 30 848 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 849 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 850 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 851 0.0 0.2 139732 1220 ? S kamailio -m 100 -M 30 852 0.0 0.2 139732 1208 ? S kamailio -m 100 -M 30 853 0.0 0.2 139736 1032 ? S kamailio -m 100 -M 30 854 0.0 0.2 139732 1176 ? S kamailio -m 100 -M 30 855 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 856 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 857 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 858 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 859 0.0 0.2 139732 1396 ? S kamailio -m 100 -M 30

2 1

Settings
by Robert Schleusener 19 Mar '12

19 Mar '12

Hello, on the iptel.org website I created an account. Also my own domain is registered and all works fine. I prefer numbers in the SIP-ID. With my own domain, I thought, I can use every number. But that is not true. Every number must start with the digit 8, for example 8555(a)mydomain.de. What is the reason for that? Have I the possibility to change that setting? I can also create aliases, for example 4123(a)mydomain.de and that works fine. But the telephonee sees still the number 8555 from the account name. Any ideas? Best regards, Robert

2 1

dns lookup and store IP in avp..
by MingHon 19 Mar '12

19 Mar '12

Hi All, is this function available?? http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati… Thanks :) -- Regards, MingHon

3 2

Hi Daniel.
by Andres Collazos 17 Mar '12

17 Mar '12

thanks for all the support for all this years. Can you please help me to know if there is any way to route sip calls based on transport protocol, for example a call incoming on tcp i will assign a route and if a call comes in udp i will assign a different route. my scenario is calls coming from different devices registering to kamailio and then kamailio send those calls to asterisk. unfortunately i have to create a different peer set for each device. for this scenario i have two types of UAs and they need completely different configuration on the asterisk switch. i am planing on segregate the traffic and build a media server for each type of device means 2 asterisk, teh only way that i can identify those incoming registrations is by the use of the port one client connects trough udp the other trough tcp. I appreciated any input in this matter. and again thank a lot to all for the great support. Andres Collazos.

2 1

Presence in Kamailio
by Faisal Rehman 16 Mar '12

16 Mar '12

Hi, I am building Kamailio as a Stand-alone XMPP server so that XMPP clients can IM each other; but I also want the below functionality: The XMPP clients should all be able to see each others presence So do I need to look out the Kamailio presence module and will it fulfill my above requirement. Regards, Faisal Rehman

3 3

Again problems with ACK and Bye
by Stoyan Mihaylov 16 Mar '12

16 Mar '12

In wireshark, there are 2 fields - Source and Destination. I am curious if there are variables in Kamailio which can show me those values. System: MYIP - IP of kamailio sip.mydomain.com - address of kamailio User (Imsdroid for Android) ->Kamailio->Asterisk->Kamailio (same) ->Other user. Problem - register, voice, video, and ring are OK, but ACK and BYE are going in dead loop - and this happens may be around in 50% of cases - not in all cases. If I am using other client (like hardware SIP phones), then I have no problems, but it is really hard to find some client for Android which supports video with Kamailio. To solve problem, I made one function in Kamailio.cfg - ACKBYE, which is called only within dialog. And within dialog is called in main route. route[ACKBYE] { #!ifdef WITH_MYFORWARD xlog("SCRIPT: AckBye $td;$si;$du;$dd\n"); if((($td=="sip.mydomain.com")||($td=="MYIP"))&&($si=="MYIP")){ $du=$sht(forw=>$ft); xlog("SCRIPT: AckBye2 $td;$si;$du\n"); return; } #!endif return; } Of cource I have if(is_method("INVITE")){ ds_select_dst("1","4"); $sht(forw=>$ft)=$du; } This worked for long time. But then we updated client (libraries etc), and problem appeared again - from time to time. Calls between 2 same clients can pass 2 or 5 times, and then problem appear. Before - $td and $si showed destination and source but now there is difference: ERROR: <script>: SCRIPT: AckBye 192.168.2.251;MYIP;<null>;<null> $td in this case show destination, which is correct, but in wireshark, destination and source = MYIP. ACK sip:MYIP;lr=on;nat=yes SIP/2.0 Via: SIP/2.0/UDP MYIP;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP MYIP;rport=5060;branch=z9hG4bKcydzigwkX ............................................ Via: SIP/2.0/UDP MYIP;rport=5060;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.2.105:44794;branch=z9hG4bK338581244;rport=44794 From: <sip:11171@sip.MYDOMAIN.com>;tag=1687441663 To: <sip:11175@192.168.2.251>;tag=as13c57096 Contact: <sip:11171@MYIP :5060;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: 4026cd88-aab9-19db-dfbe-1510baafdbb3 CSeq: 473985105 ACK Content-Length: 0 Max-Forwards: 0 Proxy-Authorization: Digest username="11171",realm="sip.stribogkonsult.com ",nonce="T2NZUU9jWCWxMUHSQweHXVQppHS8iFhd",uri="sip:11175@MYIP :5060",response="c5a95942af1502c208fa05d34c405907",algorithm=MD5 Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000 User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.0.453 (doubango r653)

2 1

Re: [SR-Users] define-check for dlg_var doesn't work
by Daniel-Constantin Mierla 16 Mar '12

16 Mar '12

On 3/16/12 10:25 AM, Sven Knoblich wrote: > Thanks Daniel, > this solves the problem. I will fix it for master and cherry pick it > for 3.2 as well. ok -- keeping the list cc-ed will allow people reading the thread on users to see the final results. Cheers, Daniel > > Bye Sven > > > Am 15.03.2012 23:16, schrieb Daniel-Constantin Mierla: >> Hello, >> >> from the code, I see that if the value does not exists, the get-function >> for dlg_var is just returning 0. Probably it should return PV null >> value. >> >> Try again after changing return 0 in return pv_get_null(...) inside >> function pv_get_dlg_variable(...) from dlg_var.c >> >> If works, you can commit the fix. >> >> Cheers, >> Daniel >> >> On 3/15/12 4:53 PM, Sven Knoblich wrote: >>> sip-router:master >>> >>> Hi all, >>> i want to use the define-check also for dlg_var. But after testing a >>> bit it seems to me, that define does not work with dlg_var's. I am >>> always getting a true. Did i miss something? >>> >>> # $dlg_var(test) is not defined >>> >>> if( defined $dlg_var(test)) >>> { >>> xlog("L_Notice","dlg_var $dlg_var(test) is defined\n"); >>> } >>> >>> I hope anybody can help me to clarify this. >>> >>> Thanks in advance, >>> Sven >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

2 2

define-check for dlg_var doesn't work
by Sven Knoblich 16 Mar '12

16 Mar '12

sip-router:master Hi all, i want to use the define-check also for dlg_var. But after testing a bit it seems to me, that define does not work with dlg_var's. I am always getting a true. Did i miss something? # $dlg_var(test) is not defined if( defined $dlg_var(test)) { xlog("L_Notice","dlg_var $dlg_var(test) is defined\n"); } I hope anybody can help me to clarify this. Thanks in advance, Sven

2 3

ser.cofig
by Marwan Idriss 16 Mar '12

16 Mar '12

Greeting, I want to do the following : if (method=="REGISTER") { then save the message received in the ser database } Regards Marwan

2 1

Needed Modules Functions
by Marwan Idriss 16 Mar '12

16 Mar '12

Greeting, I need a detailed modules explanation and their function Regards Marwan

2 1

[nube] Simple TCP/TLS proxy?
by Matthew Roderick 15 Mar '12

15 Mar '12

Hi All, I've been googling for a little while now and although I found others asking the question and answers says "yes it's possible" I haven't managed to find out *how* to do it! I have a relatively simple scenario where I'd like to use Kamailio (or something else?) to take TCP or UDP connections from my Linksys PAP2T ATA and proxy them to my sip provide which only allows TLS connections (obviously the Linksys doesn't support TLS). The kicker is that, although I understand the theory and principles of SIP, I'm completely new to SIP software and how to configure it, etc. If anyone can point me to a (good and recent) beginners guide to Kamailio installation and configuration that would also help ;-) Many thanks Matt PS. Installation status is I have Kamailio installed on Ubuntu with TLS configured (define WITH_TLS & module installed), started and listening! I've but not yet enabled mysql (not sure if it's needed for what I want to do!)

2 1

collecting statistics
by Klaus Darilion 15 Mar '12

15 Mar '12

Hi! I want to collect statistics on a load-balancer, e.g. how many NOTIFYs, SUBSCRIBEs, ... and the respective response codes, e.g. NOTIFYs-200, NOTIFYs-408,...., response time from the backend-servers, ... Further I want to do this separately for each backend-server in the cluster, i.e. for every IP address in the dispatcher list. Built in statistics or not detailed enough, so thought of implementing the counting and measuring in script using htable and a periodic timer which dumps the htable values to log file. Thus, I will end up with lots of code, especially for dumping I have to address all possible keys manually: e.g. request route: $var(key) = $rm + "::" + $dd; $sht(stat=>$var(key)) = $sht(stat=>$var(key)) + 1; e.g. reply route: if ($rs != "200" && $rs != "480" ) { $var(key) = $rm + "::" + $si + "::XXX"; } else { $var(key) = $rm + "::" + $si + "::" + $rs; } $sht(stat=>$var(key)) = $sht(stat=>$var(key)) + 1; e.g. dumping: xlog("PUBLISH::83.136.32.169=$sht(stat=>PUBLISH::83.136.32.169)"); $sht(stat=>PUBLISH::83.136.32.169)=0; xlog("PUBLISH_200::83.136.32.169=$sht(stat=>PUBLISH_200::83.136.32.169)"); $sht(stat=>PUBLISH_200::83.136.32.169)=0;xlog("PUBLISH_480::83.136.32.169=$sht(stat=>PUBLISH_480::83.136.32.169)"); $sht(stat=>PUBLISH_480::83.136.32.169)=0; xlog("PUBLISH_XXX::83.136.32.169=$sht(stat=>PUBLISH_XXX::83.136.32.169)"); $sht(stat=>PUBLISH_XXX::83.136.32.169)=0; It works, but is not very nice, IMO. Thus, I wonder if someone has a better, more elegant way to achieve this. Thanks Klaus

2 3

MI Command in Script
by Reda Aouad 15 Mar '12

15 Mar '12

Hi, Can we call MI commands from the script ? I'm thinking about reloading dispatcher or hash table from database for example.. Thanks Reda

2 2

Kamailio consumes memory
by Martin Tichatschek 15 Mar '12

15 Mar '12

Hi everyone, I have a problem with Kamailio consuming memory. I started testing with Kamailio 3.2.1 upgraded to 3.2.2 and also tested 3.1.5, everywhere the same problem. Whereas Kamailio 1.5.0 is working fine. Kamailio -V shows the following: kamailio@test:~$ kamailio -V version: kamailio 3.2.2 (i386/linux) 98ba92 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 98ba92 compiled on 17:51:30 Feb 29 2012 with gcc 4.2.4 kamailio@test:~$ My test scenario consists of two SIPp servers (a UAC and a UAS) and the Kamailio in the middle proxying the SIP traffic. The UAC sends 300 calls per second (INVITE, 100, 180, 200, ACK) and releases the calls (BYE, 200) after 20 sec call duration. UAS is simply sending back the right responds. Protocol is UDP. After some minutes the Kamailio floods the syslog file with error messages indicating that he is running out of memory. The SIPp shows retransmissions and also timeouts. When I stop the test and do a single call I get some lines of error messages. I found three different sets of error messages for a single call so far: Mar 2 09:35:44 test P-CSCFtest[21251]: ERROR:<core> [parser/msg_parser.c:172]: get_hdr_field: out of memory Mar 2 09:35:44 test P-CSCFtest[21251]: INFO:<core> [parser/msg_parser.c:353]: ERROR: bad header field [To:<sip:00498111111] Mar 2 09:35:44 test P-CSCFtest[21251]: ERROR: tm [t_lookup.c:1118]: ERROR: reply cannot be parsed Mar 2 09:35:44 test P-CSCFtest[21251]: ERROR:<core> [msg_translator.c:1884]: ERROR: build_res_buf_from_sip_res: out of mem Mar 2 09:35:44 test P-CSCFtest[21251]: ERROR:<core> [forward.c:794]: ERROR: forward_reply: building failed Mar 2 09:35:44 test P-CSCFtest[21228]: ERROR:<core> [msg_translator.c:1884]: ERROR: build_res_buf_from_sip_res: out of mem Mar 2 09:35:44 test P-CSCFtest[21228]: ERROR: tm [t_reply.c:1786]: ERROR: relay_reply: no mem for outbound reply buffer Mar 2 09:35:44 test P-CSCFtest[21228]: ERROR:<core> [msg_translator.c:2036]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 311 Mar 2 10:19:43 test P-CSCFtest[21346]: ERROR:<core> [msg_translator.c:1884]: ERROR: build_res_buf_from_sip_res: out of mem Mar 2 10:19:43 test P-CSCFtest[21346]: ERROR: tm [t_reply.c:1786]: ERROR: relay_reply: no mem for outbound reply buffer Mar 2 10:19:43 test P-CSCFtest[21346]: ERROR:<core> [msg_translator.c:2036]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 311 Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [action.c:852]: ERROR: do_action: memory allocation failure Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [action.c:1568]: run action error at: /usr/local/etc/kamailio//kamailio.cfg:28 Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [action.c:852]: ERROR: do_action: memory allocation failure Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [action.c:1568]: run action error at: /usr/local/etc/kamailio//kamailio.cfg:32 Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR: tm [t_lookup.c:1326]: ERROR: new_t: no valid From in INVITE Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR: tm [t_lookup.c:1478]: ERROR: t_newtran: new_t failed Mar 2 10:20:16 test P-CSCFtest[21328]: WARNING:<core> [receive.c:209]: WARNING: receive_msg: error while trying script I repeated the test various times, it always generates a memory problem. The "top" command shows that the free memory is decreasing constantly during the test. Stopping the test and waiting for some time (1 hour) doesn't increase the free memory showed by "top". Only a restart of the Kamailio frees the memory again. I executed the test with 30 calls per second too and got the same behaviour (after some more minutes). I finally simplified the Kamailio config as far as possible and came to this minimum: #!KAMAILIO listen=10.10.10.10:5060 children=30 user="kamailio" group="kamailio" debug=2 disable_core_dump=yes log_facility=LOG_LOCAL0 log_name="P-CSCFtest" server_signature=no sip_warning=0 syn_branch=0 disable_tcp=yes ####### Modules Section ####### mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" loadmodule "tm.so" ####### Main Routing Logic ######## route{ if (src_ip==1.2.3.4) { rewritehost("5.6.7.8"); } else { rewritehost("1.2.3.4"); }; rewriteport("5061"); t_relay(); } This config still generates the memory problem. After replacing the "t_relay()" with the stateless "forward()" the problem is gone. I played around with the number of child processes too, but no difference. Is this a memory leak of the TM module or is there a different explanation? Regards Fred

3 5

Timer
by Ravi DHIRAJLAL 15 Mar '12

15 Mar '12

Hello we installed kamailio succesfully but we have a few problems regarding calls towards special numbers in which an INVITE is not replied with 200 OK but instead with 183 session progress sent at 1 min interval, during which the caller hears a waiting message. unless a human operator answer the call, no 200 OK is replied. if that waiting time is too long, we noticed that the call is hanged up after 3 min, 180s, that is 3 * session progress . 408 timeout is sent to the caller and cancel to the callee. we tried to adjust many timers listed in the tm module, but not seems to work we'd like to see if anybody has a clue about that ? thank you very much

2 6

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