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[Kamailio-Users] Need Assistance Tracking Down A SIP Signaling Problem
by Geoffrey Mina 25 Sep '09

25 Sep '09

Hello, I am wondering if anyone can help me determine what the problem with my SIP signaling is here. I have attached two Excel files (can provide other formats if someone doesn't have Excel). The environments in both scenarios are almost identical... although I am dealing with two different carriers upstream. In my environment, I have Kamailio (1.5) sitting in front of a multitude of Asterisk machines. I am using the dispatcher module to distribute INVITE requests across the network. I am doing some interop with a new carrier and we are struggling a bit with some looping scenarios. They are sending invites to my Kamailio server, I am forwarding to asterisk. The two XLS files (Success.xls) and (Fail.xls) have sip_trace information for the transactions. As you can see in the Fail.xls, the ACK to the 200 OK gets looped back to the Kamailio server and never ends up reaching the Asterisk machine. In the Success.xls, this doesn't happen. The only thing I can see that's different is the Success carrier appears to be replacing the RURI in the ACK with the value of the Contact header... in the failed one, the RURI of the ACK is the IP of my Kamailio gateway. I have read through the RFC 3261 to try and figure out what is breaking this, but haven't been able to. I basically need to determine if the signalling is correct... what is wrong with my config... and if the signalling is wrong, what the carrier is doing incorrectly. Thanks in advance! Geoff

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[Kamailio-Users] Diameter
by Ghaith ALKAYYEM 24 Sep '09

24 Sep '09

Hello list, Does anybody have any idea or recommendation about integrating diameter with SER? As far as you know the available module is obsolete, i'm trying to implement the module "CDP" available with OpenIMS: http://www.openimscore.org/docs/ser_ims/CDP.html Thank you

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[Kamailio-Users] Unable to lookup host in c= line,
by Tincho ylm 24 Sep '09

24 Sep '09

Hi all! I'm trying to make a realtime environment openser-asterisk. At this moment, everything is working fine but I have problems with VM feature. What I'm doing is: A call B If B is busy and returns 486 Openser send this to Asterisk. failure_route[1] { if(!t_was_cancelled()) { if (t_check_status("487|486")) { rewritehostport("200.xx.xx.xx:5060"); append_branch(); xlog("L_INFO","---> To Voicemail...\n"); setflag(10); route(1); exit; } if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } } Asterisk cant handle it because this error: chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' I'm using: Kamailio 1.4.4 Rtpproxy Asterisk 1.4.22 Is it a rtpproxy issue? it can fix it? If I make a ethereal capture, I can see in the MESSAGE BODY this: connection inrfomation (c): IN IP4 200.xx.xx.xx200.xx.xx.xx What can I do? thanks in advance.

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[Serusers] Search trough status line for specific reason phrase
by Josip Djuricic 23 Sep '09

23 Sep '09

Dear, is there any way to search inside Status-Line of response message? I have strange b2bua that always returns SIP 500 just with different reason phrase. I need to match on them so that I can rewrite failure messages. I thought about something like this: failure_route[1] { if (t_check_status("500")) { #here I need to search trough status line for specific reasons, but have no idea how to do that if (search("^Specific message")) { t_reply("480", "Temporarily Unavailable"); break; } } } Thank you J.

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[Kamailio-Users] H323 - SIP Signalling gateway for Kamailio
by General Lee 23 Sep '09

23 Sep '09

Hi, I'm looking for a H323 - SIP signalling gateway which I can use to bridge video calls between my H323 Gatekeeper and Kamailio system. Has anyone done anything like this before? Are you able to suggest any suitable h323 - SIP video gateway solutions I could use? Thanks -- View this message in context: http://www.nabble.com/H323---SIP-Signalling-gateway-for-Kamailio-tp25530990… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] SER as a stateless proxy server
by Muhammad Ali 23 Sep '09

23 Sep '09

Hi, How SER can be configured as a stateless proxy server? Best regards M Ali

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[Kamailio-Users] content about kamailio start error
by zhangchao00001 23 Sep '09

23 Sep '09

Hello everyone, i have some questions about kamailio. After i install rtpproxy 0.3version, the kamailio server can't work any more. The following is the error message. Dose any one know the reason? dose the rtpproxy 0.3 change something that kamailio need? Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module registrar Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module textops Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module uri_db Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module siputils Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module xlog Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module acc Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:57 [4925] ERROR:core:build_req_buf_from_sip_req: out of pkg memory Sep 23 11:07:57 [4925] ERROR:tm:print_uac_request: no more shm_mem Sep 23 11:07:57 [4925] ERROR:tm:t_forward_nonack: failure to add branches Sep 23 11:07:57 [4925] ERROR:tm:_reply_light: failed to allocate shmem buffer Sep 23 11:07:57 [4925] ERROR:tm:relay_reply: no more share memory Sep 23 11:07:57 [4925] ERROR:tm:_reply_light: failed to allocate shmem buffer

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[Kamailio-Users] Comparing AVPs to user variables
by Alex Balashov 23 Sep '09

23 Sep '09

Is there a known bug in Kamailio 1.4.1 that causes comparisons like this to be inaccurate? get_profile_size("people", "$avp(s:person_id)", "$var(u_call_count)"); ... if($var(u_call_count) >= $avp(s:call_limit)) In this case, $var(u_call_count) was 3 and $avp(s:call_limit) was 26 - I printed both values beforehand. Yet, the comparison was evaluated as true. I worked around the problem by storing the profile size in an AVP as well and comparing them instead, and it seems to have solved the problem: if($avp(s:u_call_count) >= $avp(s:call_limit)) However, I was wondering if this is a known problem in 1.4.1 and possibly 1.4.x. Thanks! -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Kamailio-Users] Stupid softswitch sending a CANCEL with "To tag" in parallel forking
by Iñaki Baz Castillo 22 Sep '09

22 Sep '09

Hi, I've experimenting an absurd issue when a softswitch calls to Kamailio and receives two 180 (with different To tags) due to parallel forking made by Kamailio. When the softswitch receives the second 180 (with the second To tag) it sends a ridiculous CANCEL with To tag matching the second To tag received (yes, a CANCEL with To tag !!!). Of course Kamailio replies "403 Why do you send a CANCEL with To tag? please read the RFC 3261". Unfortunatelly the softswitch didn't go to the school and ignores the 403 by repeating the CANCEL. And the worst, when *any* of the ringing phones answerw, the softswitch ignores the 200 so the call is never established. IMHO there is nothing I can do to "fix" this stupid issue. I've tried to reply 200 for the CANCEL but the softswitch is not happy (it seems that it expects the 487 from the canceled branch). And of course, if I do "t_relay()" for the CANCEL then Kamailio cancels both branches (the correct behavior). The only hyper exotic "solution" I can imagine is the following: - Kamailio receives the wrong CANCEL with To tag "2". - Reply 200 for the CANCEL to make the softswitch happy. - Reply 487 with To tag "2" to make the softswitch happy (IMHO this is not possible). - Generate a spoofed BYE for the second branch (that with To tag "2"). Of course, this is not possible with dialog module as it doesn't handle parallel forking. Any suggestion? (yes, I would like to kill the softswitch). Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] new feature: update sip msg content in request route
by Daniel-Constantin Mierla 21 Sep '09

21 Sep '09

Hello, I just committed a new function in modules_k/textops - msg_apply_changes(). Hopefully the name is suggestive enough - the function applies the changes done to the content of sip message - e.g., if you added a new header, once you use the function, that header will become visible in config file. Therefore the function tries to overcome the overhead of looping various messages back to sip server just because modifications were not visible once done. Another benefit is ability to access latest value for a field (being header or body content) in case of need to hashes or signatures for security purposes. The function can be used only in request route for now. Be carefully when using it, since it changes the expectation you had so far of always working on initially received sip message. Once the function is sued, initial content is lost forever. Since it builds a new buffer and re-parses, it is not very recommended to use it extensively. Example of usage: append_hf("My-Header: yes\r\n"); if(msg_apply_changes()) { # msg buffer has a new content if(is_present_hf("My-Header")) { # will get always here } } Testing and feedback is welcome! Cheers, Daniel

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