sr-users May 2010

sr-users@lists.kamailio.org

67 participants
106 discussions

Re: [SR-Users] no Contact in REFER generated with dlg_bridge
by Anton Roman 08 Jul '10

08 Jul '10

Hi, regarding the missing Contact header in the REFER message, it can be solved by including some lines in /modules_k/dialog/dlg_ transfer.c. In this file, the Contact of the initial INVITE generated with dlg_bridge command can be easily changed as well. The problem now is only in the processing of the ACK generated by the TM module. When this ACK leaves the Kamailio server the branch value of the topmost Via header is '0'. Below you can find the debug output where the ACK is generated and routed by Kamailio. Any idea about where the error can be? thanks in advance, regards Anton 11(5056) DEBUG: tm [t_msgbuilder.c:791]: building ACK for out-of-dialog INVITE (using RS in RR set). 11(5056) DEBUG: tm [t_msgbuilder.c:967]: ACK RURI: `sip:235@10.1.3.15:5061', NH: `sip:10.1.2.122;lr=on'. 11(5056) DEBUG: tm [t_reply.c:1059]: ->>>>>>>>> T_code=180, new_code=200 11(5056) DEBUG: tm [t_reply.c:1794]: DEBUG: local_reply: branch=0, save=0, winner=0 11(5056) DEBUG: tm [t_reply.c:356]: DEBUG: update_totag_set: new totag 11(5056) DEBUG: tm [t_reply.c:1831]: DEBUG: local transaction completed 11(5056) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x7f9767eb1268, callback type 256, id 0 entered 11(5056) DEBUG: dialog [dlg_transfer.c:207]: completed with status 200 11(5056) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd 9(5052) DEBUG: <core> [parser/msg_parser.c:612]: SIP Request: 9(5052) DEBUG: <core> [parser/msg_parser.c:614]: method: <ACK> 9(5052) DEBUG: <core> [parser/msg_parser.c:616]: uri: < sip:235@10.1.3.15:5061> 9(5052) DEBUG: <core> [parser/msg_parser.c:618]: version: <SIP/2.0> 9(5052) DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bKf7bc.00856364.0>; state=16 9(5052) DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 9(5052) DEBUG: <core> [parser/msg_parser.c:500]: parse_headers: Via found, flags=2 9(5052) DEBUG: <core> [parser/msg_parser.c:502]: parse_headers: this is the first via 9(5052) DEBUG: <core> [receive.c:137]: After parse_msg... 9(5052) DEBUG: <core> [receive.c:177]: preparing to run routing scripts... 9(5052) DEBUG: sl [sl_funcs.c:335]: to late to be a local ACK! 9(5052) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=94cf927f9c6eed24i1 9(5052) DEBUG: <core> [parser/parse_to.c:808]: end of header reached, state=29 9(5052) DEBUG: <core> [parser/msg_parser.c:174]: DEBUG: get_hdr_field: <To> [43]; uri=[sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>] 9(5052) DEBUG: <core> [parser/msg_parser.c:176]: DEBUG: to body [ sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>] 9(5052) DEBUG: <core> [parser/msg_parser.c:154]: get_hdr_field: cseq <CSeq>: <10> <ACK> 9(5052) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_body : content_length=0 9(5052) DEBUG: <core> [parser/msg_parser.c:90]: found end of header 9(5052) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 9(5052) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd 9(5052) DEBUG: <core> [parser/parse_to.c:808]: end of header reached, state=29 9(5052) DEBUG: sanity [mod_sanity.c:220]: all sanity checks passed 9(5052) ERROR: <script>: kamailio!!!: ACK 9(5052) DEBUG: siputils [checks.c:68]: totag found 9(5052) DEBUG: rr [loose.c:85]: is_preloaded: No 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==10 && [10.1.3.15] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==10 && [10.1.3.15] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [forward.c:356]: check_self: host != me 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==9 && [10.1.2.122] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==9 && [10.1.2.122] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==10 && [10.1.2.122] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: rr [loose.c:781]: Topmost route URI: 'sip:10.1.2.122;lr=on' is me 9(5052) DEBUG: <core> [parser/msg_parser.c:90]: found end of header 9(5052) DEBUG: rr [loose.c:257]: No next Route HF found 9(5052) DEBUG: rr [loose.c:800]: No next URI found 9(5052) DEBUG: rr [rr_cb.c:97]: callback id 0 entered with <lr=on> 9(5052) DEBUG: dialog [dlg_handlers.c:797]: Route param 'did' not found 9(5052) DEBUG: tm [t_lookup.c:1387]: DEBUG: t_newtran: msg id=10 , global msg id=9 , T on entrance=0xffffffffffffffff 9(5052) DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=52095, isACK=1 9(5052) DEBUG: tm [t_lookup.c:720]: DEBUG: t_lookup_request: e2e proxy ACK found 9(5052) DEBUG: tm [t_reply.c:276]: DEBUG: totag for e2e ACK found: 0 9(5052) DEBUG: tm [t_funcs.c:315]: SER: forwarding ACK statelessly 9(5052) DEBUG: <core> [msg_translator.c:200]: check_via_address(10.1.2.122, 10.1.2.122, 0) 9(5052) DEBUG: <core> [forward.c:509]: Sending: ACK sip:235@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.2.122;branch=0 Via: SIP/2.0/UDP 10.1.2.122;branch=z9hG4bKf7bc.00856364.0 From: sip:click2dial@172.22.5.17 <sip%3Aclick2dial(a)172.22.5.17> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd Call-ID: 2bef8d5474ff8cd4 To: sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>;tag=94cf927f9c6eed24i1 CSeq: 10 ACK User-Agent: kamailio (3.0.1 (x86_64/linux)) Content-Length: 0

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LCR, problem with routing (prefix, from_uri, grp_id)
by Maciej Bylica 07 Jul '10

07 Jul '10

Hi ALL, I have a problem to configure LCR module to work properly (kamailio rel. 1.5) The task in my case is to handle: 1) All calls coming with CLD number prefix 482223344 from my MGW (77.77.77.77) towards kamilio and direct them to sip world (88.88.88.88) 2) Calls with CLD starting with 48 coming from SIP world towards kamailio and direct them to my MGWs. Additionaly i need to route it on the basis of CLI number. So MGWs <---> Kamailio <---> SIPProxies (88.88.88.88) In db: +----+-----------+----------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+-----------+----------------+--------+----------+ | 1 | 48 | 48 | 1 | 1 | | 2 | 48 | 49 | 2 | 1 | | 3 | 482223344 | 77.77.77.77 | 3 | 1 | +----+-----------+----------------+--------+----------+ gw, +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ | 1 | CLI48 | 1 | 77.77.77.70 | NULL | 5060 | NULL | NULL | NULL | NULL | 150 | 0 | 0 | | 2 | CLI48backup | 1 | 77.77.77.71 | NULL | 5060 | NULL | NULL | NULL | NULL | 150 | 0 | 0 | | 3 | CLI49 | 2 | 77.77.77.75 | NULL | 5060 | NULL | NULL | NULL | NULL | 15 | 0 | 0 | | 4 | CLI49backup | 2 | 77.77.77.76 | NULL | 5060 | NULL | NULL | NULL | NULL | 15 | 0 | 0 | | 5 | SIPWORLD | 3 | 88.88.88.88 | NULL | 5060 | NULL | NULL | NULL | NULL | 250 | 0 | 0 | +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ Unfortunately: - call is originated on 77.77.77.77 with CLD 48222334455 number and kamailio forward this call to grp_id=3 SIPWORLD which is okay. The problem is that if the call fail, kamailio will try to use 77.77.77.70 and 77.77.77.71 from grp_id=1 which is wrong. I have no idea how to provide a kind of huntstop in grp_id=3. I've been trying with +----+-----------+----------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+-----------+----------------+--------+----------+ | 1 | 48 | 88.88.88.88 | 1 | 1 | | 2 | 48 | 88.88.88.88 | 2 | 1 | | 3 | 482223344 | 77.77.77.77 | 3 | 1 | +----+-----------+----------------+--------+----------+ but in this case i am unable to provide kamailio with CLI number routing. Here is a part of my config file... if (!load_gws()) { sl_send_reply("503", "Unable to load gateways"); exit; } if (!next_gw()) { sl_send_reply("503", "Unable to find a gateway"); exit; } route(1); exit; and failover for route(1)... failure_route[11] { # In case of failure, send it to an alternative route: if (t_check_status("408|404|5[0-9][0-9]")) { if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); exit; } else { t_on_failure("11"); t_relay(); } exit; } } Anybody could help my to get out of that? Thx, Maciej.

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GSoC2010: welcome Marius Bucur
by Daniel-Constantin Mierla 22 Jun '10

22 Jun '10

Hello, GSoC2010 selected projects were announced several days ago. I want to welcome Marius Bucur, students at Computer Science Faculty in Bucharest. He is going to implement our proposed project about presence support for audio conference mixers: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 It will be tight related to implementation in SEMS conference mixer: http://www.sip-communicator.org/index.php/GSOC2010/SEMSAudioLevels Shortly, is about getting presence updates about participants in a conference room, who is speaking, audio levels of speakers, etc. SIP Communicator is a softphone that has support for this features, see a presentation about it at: http://www.kamailio.org/events/2010-FOSDEM/SIP.Communicator-FOSDEM-2010.pdf http://www.sip-communicator.org/ Please provide Marius hints and guidelines as fast as you can for him to get quickly into application architecture. Along with me, Marius Zbihlei will officially help in mentoring, if you have questions about GSoC project administration, etc., contact one of us. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

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t_on_failure to named failure route
by Steven C. Blair 18 Jun '10

18 Jun '10

I'm attempting to run an existing SER config file under a recent build of sip-router. I get a syntax error on the t_on_failure("noroute") call that exists in my config file. The specific error is "bad expression: type mismatch (str instead of int)". I presume this means sip-router does not support named failure route labels. Is that correct? Thanks,Steve --- ISC Networking & Telecommunications 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 215-573-8396 215-898-9348 (fax)

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call routing using kamailio 3.0.2
by Meftah Tayeb 02 Jun '10

02 Jun '10

hello, how do i route calls from local registered users to a pstn gateway? thank you

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Audio Conference: Kamailio and FreeSWTICH - Wed, June 2, 2010
by Daniel-Constantin Mierla 02 Jun '10

02 Jun '10

Hello, next Wednesday (June 2, 2010), at 18:00UTC, I will participate to FreeSWITCH weekly conference call, talking about using Kamailio and FreeSWITCH together. The draft of agenda is: * Goals of Kamailio, how it differentiates from FreeSWITCH and why using them together creates a very powerful framework to build large VoIP systems. * Kamailio for sip routing and FreeSWITCH as media server (conferencing, voicemail, IVR, announcements) * FreeSWITCH as a B2BUA for Kamailio (topology hiding, transcoding, call interrupt detection) * FreeSWITCH as prepaid engine for Kamailio * Load balancing FreeSWITCH servers with Kamailio Is an open debate, join and ask your questions. You can dial in via SIP, Skype, PSTN, see access details at: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_02 Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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Kamailio and FreeSWITCH as media server
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, a new tutorial about using Kamailio as SIP server and FreeSWITCH as media server is published at: http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms Hope is useful for some of you, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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New way of Kamailio and Asterisk Real Time Integration
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, several days ago, I published a new tutorial about Kamailio and Asterisk realtime integration, using a different approach than in the previous tutorials on this topic I wrote. Specifically, this time the Asterisk database structure is used. You can find it at: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Hope is useful for some of you, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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Two presentations at Amoocon 2010
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, next weekend (June 4-6, 2010) in Rosotck, Germany, takes place 2010 edition of Amoocon. http://www.amoocon.com I will give two talks about Kamailio - SIP Router: - Asynchronous SIP Routing - Develop Your SIP Routing in Lua More details are posted on news section of the web sites: http://www.kamailio.org/w/2010/05/kamailio-at-amoocon-2010/ If you are around and want to meet, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Re: [SR-Users] Dialplan modinit failed
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, please use the sr-users(a)lists.sip-router.org to post questions about ser or kamailio - apart of first one, most private messages are discarded. There are many people that can answer, faster and better. In this particular case, seems you haven't loaded pv module (located in modules_k). Cheers, Daniel On 5/30/10 9:50 AM, Dmitry Kirillov wrote: > Hi > > i tried to start SIP-Router (ser-3.0.2_src_2010-05-29) by including > dialplan.so in the list of loaded > modules and empty dialplan modparam. when i start sip-router, i get > > 0(23147) ERROR: <core> [pvapi.c:501]: error searching pvar "ruri.user" > 0(23147) ERROR: <core> [pvapi.c:705]: wrong char [r/114] in > [$ruri.user] at [9 (0)] > 0(23147) ERROR: dialplan [dialplan.c:163]: input pv is invalid > 0(23147) ERROR: <core> [sr_module.c:874]: init_mod(): Error while > initializing module dialplan > ERROR: error while initializing modules > > With modparam("dialplan", "attrs_pvar", "$avp(s:dest)"): > > 0(23158) ERROR: <core> [pvapi.c:501]: error searching pvar "avp" > 0(23158) ERROR: <core> [pvapi.c:705]: wrong char [s/115] in > [$avp(s:dest)] at [5 (5)] > 0(23158) ERROR: dialplan [dialplan.c:149]: invalid pvar name > 0(23158) ERROR: <core> [sr_module.c:874]: init_mod(): Error while > initializing module dialplan > ERROR: error while initializing modules > > there is no dialplan function call in ser.cfg. > > Please, help > -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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sr-users May 2010