sr-users May 2012

sr-users@lists.kamailio.org

84 participants
140 discussions

Config include
by Konstantin M. 09 May '12

09 May '12

Hi, I would like (and a many people here I believe) to have a functional of including a multiple config files like (foe example asterisk's #include "path/to/some/config.conf"). Is it possible to implement a such feature ? Thanks!

4 7

Rtpproxy issue with connection information
by Openser Kamailio 09 May '12

09 May '12

Hi, i'm currently working with kamailio 3.2 and rtpproxy 1.2.1. Both are set up on the same computer. When rtpproxy adds an SDP to an Invite, it adds two IPv4 addresses in owner/creator session and connection information field with an error, i.e: *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 * 172.27.170.984* 172.27.170.98 *Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98 the correct ones should be : *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 172.27.170.98 *Connection Information (c)*: IN IP4 172.27.170.98 Thanks for help!!

3 3

Alias configuration with multidomain
by Arjan Kuiken 09 May '12

09 May '12

Hello SR-users, I want to configure a kamailio 3.2.3. We want to use it to connect multiple SIP servers as users and let the Kamailio work as a Trunk/ Proxy connected to a PSTN gateway. We have multiple domains and aliases per SIP servers. When I register a SIP server to Kamailio everything goes well, but when I try to call an dbalias it doesn't work. It gives 2 errors (trace included). I think something is done wrong in my kamailio.cfg but I don't have a clue anymore. I included the cfg file as attachement. Can you please have a look? Thanks in advance! Kind regards, Arjan Kuiken Technical Specialist [intercity-zakelijk-logo-handetekening (2)] Kruisweg 659, 2132 NC, Hoofddorp T: 020 - 655 3000 F: 020 - 6531534 E: Arjan.kuiken(a)intercityzakelijk.nl<mailto:arjan.kuiken@intercityzakelijk.nl> I: www.intercityzakelijk.nl<http://www.intercityzakelijk.nl/>

2 2

Accounting '180 Ringing' without SDP
by Stephen Dodge (Bistech) 09 May '12

09 May '12

Hello, We are running Kamailio 3.1.5 and using the acc module to generate CDRs and provide reporting information. We have a requirement to report ring time and we do this by accounting on early media for 183 provisional responses. Unfortunately we have a couple of customer PBX's that cannot provide a SDP with the ringing response. i.e they just respond with a '180 ringing'. My understanding of the module is that the "early_media" parameter is looking for a SDP in either a 180 or 183 response before accounting. Although it may be possible to account the 180 without SDP using the acc module I cannot find anything obvious in the documentation. If it is possible could someone point me in the right direction or suggest a creative solution, otherwise can I ask that this functionality be introduced. Many Thanks. Steve Dodge. ________________________________ 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.

2 1

detect retransmission messages...
by Vineet Menon 09 May '12

09 May '12

Hi, How do we detect a message as retransmitted message, in a transaction oriented proxy? Regards, Vineet Menon

2 1

Re: [SR-Users] detect retransmission messages...
by Alex Balashov 09 May '12

09 May '12

t_check_trans() generally does the trick. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Vineet Menon <mvineetmenon(a)gmail.com> wrote: >Hi, > >How do we detect a message as retransmitted message, in a transaction >oriented proxy? > >Regards, > >Vineet Menon > >_______________________________________________ >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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kamdbctl
by Klaus Darilion 09 May '12

09 May '12

Hi! I wonder what is the status of kamdbctl tool for 'migrate' command - should it work? The description says: kamdbctl migrate <old_db> <new_db> .....................(migrates DB from 1.2 to 1.3, not implemented yet!) This seems a bit outdated. regards Klaus

3 3

kamdbctl error
by Vineet Menon 09 May '12

09 May '12

Hi, I am compiling sources of kamailio 3.2.3 on ubuntu 12.04. I have used make command > `make group_include="standard mysql" all` > to make the sources. and then > `make install`..... > Now when i do kamdbctl it throws an error, $ kamdbctl create > ERROR: could not load the script in > /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL > ERROR: database engine not loaded - tried 'MYSQL' > Any guess what is the problem??? Regards, Vineet Menon

3 7

Kamailio [No Audio]
by Fred Flintsone 09 May '12

09 May '12

I am attempting to route local registered users to local registered users without going to the media server. I have a media server [*] for PSTN. But it doesn't support all the CODECS i wan't to use. Signalling seems fine, but i get NO AUDIO on either call leg. I have the RTP captures from the local side making the call and the media packets seem to be going exactly where the SDP dictates. So i'm a little confused. Is this where an RTP Proxy would come in handy? I haven't been able to get the audio to work without going to the media server. If an RTP Proxy is the answer, how much overhead does the proxy add to the Kamailio server? Is it something that you don't run on the same machine and use a distributed environment of RTP proxies on other servers? Or is this something that should be fine and working without a media proxy? I really appreciate any help! Thank you, Fred

2 1

Bug: Pua_reginfo
by Gnaneshwar Gatla 08 May '12

08 May '12

Hello all, There seems to be an deadlock between registrar and pua_reginfo module whenever a SIP client refreshes its registration. 1. The pua_reginfo does not publish Event:reg when a sip client registers for the first time. 2. The pua_reginfo module does not use ul.lock_udomain() to lock the ul table to get the record. 3. The pua_reginfo does not release the record that it accessed from ul.get_urecord() Since the record is not released, when the client re-registers, the registrar module is unable to query for the userlocation and sends a 200 OK response with 0 bindings. Which leads to no contact header in the response message and eventually deletion of the record in the location table(expires). I have tried to fix this using lock_udomain, release_urecord and unlock_udomain, but failed to rectify the error. The deadlock happens in modules_k/usrloc/udomain.c (db_load_urecord()). if (ul_dbf.query(_c, keys, 0, vals, columns, (use_domain)?2:1, 13, order, &res) < 0) My Question: 1. is pua_reginfo supposed to delete record from location database and send the same data to a presence server? a. If above is true, then should the registrar module insert a new record in the location table? b. If above is false, the registrar module should be able to refresh/update the record in the location table. This error is also impacts pua_usrloc module too as it cannot access the record and sends creates an incorrect uri. Help: I'm new and learning how Kamailio works, is there a clean way that pua_reginfo module gets the record? Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

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sr-users May 2012