[SR-Users] Asterisk realtime with kamailio Load balancing issue for sip user

Hi All, We have finished the Kamailio & Asterisk real time integration and load balancing also done using dispatcher module. Queue and voice mails are load balancing as well.When we are calling extension to extension it is showing in all the servers.It seems extension are not load balancing as pe

[SR-Users] When to rewrite o= in SDP ?

Hi, I read in http://kamailio.org/docs/modules/1.4.x/nathelper.html that Kamailio might rewrite ip address in o= in SDP to facilitate NAT traversal. It may decide to use to original IP or use IP of RTP server. What factors affect its decision ? -- Khoa Pham HCMC University of Science Faculty of

[SR-Users] not able to install Kamailio

Hello thereI hope that you can help me, I have an issue installing Kamailio, when I issue th command "make all" I receive the following error resolve.h:471:4: error: #error neither gethostbyname2 or getipnodebyname present make: *** [action.o] Error 1 I am using debian squeeze, please let me kno

Re: [SR-Users] Kamailio not increasing cseq

Daniel, Any further Help you can give me? Regards, Camila *From:* Camila Troncoso [mailto:ctronc...@redvoiss.net] *Sent:* jueves, 21 de febrero de 2013 9:52 *To:* 'mico...@gmail.com'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' *Subject:* RE: [SR-

Re: [SR-Users] Cannot hear voice with symmetric NAT and STUN

Hi David, I already read all of these. But the problem seems contradict with what I read On Thu, Feb 28, 2013 at 9:56 PM, David wrote: > Hello, > > Look on Wikipedia and read the articles for SIP, RTP, NAT and STUN. > > The answer to your question can be found in the above articles. > > David >

[SR-Users] Which is first, t_on_reply or callbacks in C ?

Hello, Can someone tell me which is executed first ? the on reply route setup with t_on_reply() or the callbacks in the code that allows the module dialog to insert the confirmed dialog into the dialog table ? Thanks, David ___ SIP Express Router

Re: [SR-Users] Cannot hear voice with symmetric NAT and STUN

Hello, Look on Wikipedia and read the articles for SIP, RTP, NAT and STUN. The answer to your question can be found in the above articles. David On 13-02-27 10:26 PM, Khoa Pham wrote: Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP. I read that STUN is for non-symmetric NAT

Re: [SR-Users] Help with Asterisk RT integration

On Wednesday 12 December 2012 04:19:48 Jon Morby wrote: > For legacy reasons Asterisk needs to be in the critical path on this > particular build … what I'm looking for is a simple recipe and some > helpful pointers on how to implement it that will allow enable me to swing > (K) into the path betwe

Re: [SR-Users] UnixODBC cant insert sip_capture record on sql server

I meant the columns, not the values. Insert into sip_capture([id],,[authorization]) Values("1",."auth goes here") From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian Sent: Thursday, February 28, 2013 2:59 PM To: sr-user

[SR-Users] UnixODBC cant insert sip_capture record on sql server

Hello, We are using SQL Server 2008 as our database. When the sip_capture module tries to insert a row, the following error is given: db_unixodbc [con.c:220]: unixodbc:SQLExecDirect=42000:1:156:[FreeTDS][SQL Server]Incorrect syntax near the keyword 'authorization'. Using Kamailio 3.3.4. Is the