sr-users March 2010

sr-users@lists.kamailio.org

60 participants
117 discussions

basic sip-router installation tutorial
by Meftah Tayeb 12 Mar '10

12 Mar '10

hi all, please could anyone give me a simple step by step tutorial to setup sip-router in a debian OS with mysql and other things? from jit, no zip file... thanks

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[Kamailio-Users] Handling 3XX messages
by Bruno Machado 11 Mar '10

11 Mar '10

Hi! I have a Kamailio 1.5.2 handling 3XX messages. The commands below are executed in a failure route. If the first contact is not reachable or even a negative answer is received, the algorithm will call a second failure route, executing the 't_next_contacts'. Following this logic, the server will try several endpoints. The first 200 OK received from one this endpoints will establish the call. If all the contacts are tried, the server will finish the session with sending an error to the source. get_redirects("15:15","Redirect"); if(!t_load_contacts()) { t_reply("500", "Server Internal Error - Cannot load contacts"); exit; }; if(!t_next_contacts()) t_reply("404", "Not found"); else t_relay(); Ok... I found just one problem. If the 3XX message brings 5 different contacts, (all them with different q value), Kamailio just try 4 of them. Contact: sip:20182#556250982222@aaa.aaa.aaa.aaa;q=0.015, sip:3441#556250982222@bbb.bbb.bbb.bbb;q=0.014, sip:3441#556250982222@ccc.ccc.ccc.ccc;q=0.013, sip:50014#556250982222@ddd.ddd.ddd.ddd;q=0.012, sip:777#556250982222@eee.eee.eee.eee;q=0.011 After the 300 message, the server sends a invite to the first contact. Seconds later, analysing the debug screen, I see: DBG:tm:t_next_contacts: next contact is <sip:3441#556250982222@bbb.bbb.bbb.bbb The INVITEs are sent to this IP. After some seconds the server calls the third contact: DBG:tm:t_next_contacts: next contact is <sip:3441#556250982222@ccc.ccc.ccc.ccc And so on until this, the last contact: DBG:tm:t_next_contacts: next contact is <sip:777#556250982222@eee.eee.eee.eee ERROR:core:add_avp: 0 ID or NULL NAME AVP ERROR:tm:t_next_contacts: setting of fr_inv_timer_avp failed The proxy sent INVITEs for all them, except the last one because this failure. The function 't_next_contact' returns an error value. Somebody has a hint about this? Regards, bruno machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] Advanced Call Scenario / One Way Audio
by Brandon Armstead 11 Mar '10

11 Mar '10

**resending as I am not sure if this made it out the first time, I do not believe it did, if this is a duplicate -- my apologies.** Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 -> REGISTRAR-01 -> Kamailio-01 -> Asterisk (New Call-ID + Asterisk in Media Path) -> Kamailio-01 -> REGISTRAR-02 -> UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared "USRLOC" database to simulate calls from "PSTN" to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner "traversal" This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 -> UA2, and forces P2 -> UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this "bridge / interconnection" We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP As well as the initial Asterisk in "the middle" SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead

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[Kamailio-Users] 400 Bad Request
by Ernest Mavrel 10 Mar '10

10 Mar '10

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[Kamailio-Users] cdr_id not registered in acc
by Anders 10 Mar '10

10 Mar '10

Hi, (Kamailio 1.5 + freeradius + cdrtool) After a call is finished, it's INVITE and BYE are registered just fine in the acc table, but the records in acc never receive a cdr_id, and (therefore) nothing is inserted the cdrs table. Ideas on where to start looking? Thanks, Anders

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[Kamailio-Users] DEFINE question
by Uriel Rozenbaum 10 Mar '10

10 Mar '10

Hi guys, The question's simple, is there any pre-processor command to DEFINE constants? Thanks! Uriel

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[Kamailio-Users] OpenSER+Asterisk+BLF
by Mark Sayer 10 Mar '10

10 Mar '10

We are needing to modify the configure of a currently operating OpenSER to properly relay the SUBSCRIBE and SIP-NOTIFY messages that are sent between Asterisk and a phone that supports BLF (like the Snom 300 or Yealink T26). Our setup includes an OpenSER 1.2 & Asterisk 1.4.17 in the same box. OpenSER performs all registration, authentication and NAT. Asterisk handles the media and the accounting. In a pure Asterisk environment a "hint" would be setup in the Asterisk extensions.conf file and the phone (UA) would SUBSCRIBE to that HINT. Once Asterisk has registered that UA to the HINT then it sends SIP-NOTIFY messages to the UA as the status of the channel changes (available, ringing, busy). Our current openser.cfg file makes no mention of either SUBSCRIBE or NOTIFY which is an obvious reason that my Asterisk installation never registers the UA to the HINT. Is anyone interested in getting paid to fix this for us (we're too stupid to do it ourselves) or to offer another solution for controlling BLF in this setup. Thanks, Mark

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[Kamailio-Users] small typo in rewriteport example on Kamailio Wiki
by Andreas Heise 09 Mar '10

09 Mar '10

Hello, there is a small typo in rewriteport example on Kamailio Wiki should be rewriteport("5070"); instead of rewriteport(5070); http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#rewriteport_s… regards, Andreas

2 1

[Kamailio-Users] Kamailio (OpenSER) v3.0.1 Released
by Daniel-Constantin Mierla 08 Mar '10

08 Mar '10

Hello, the first patch release for 3.0 series is out as version 3.0.1. It includes the fixes to issues discovered since 3.0.0. Database structure and configuration file compatibility are preserved so the upgrade from 3.0.0 is straightforward. Links and more details are available at: http://www.kamailio.org/w/2010/03/kamailio-v3-0-1-released/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] [OT] BYE sip:SIMPLE&XCAP@ietf.org SIP/1.0
by Iñaki Baz Castillo 08 Mar '10

08 Mar '10

Hi, if somebody is bored and has 5-10 minutes of spare time I suggest to read the mail I've sent to sip-implementors maillist, full of hate and fury: https://lists.cs.columbia.edu/pipermail/sip-implementors/2010- March/024529.html Enjoy. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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sr-users March 2010