Re: [OpenSIPS-Users] rtpproxy not relaying data

Hi Răzvan Crainea: I was sending an U command instead of an L command. This is what I interpreted in the documentation. Is working great now. Thanks a lot! This was worrying me a lot. I've being dealing with this RTP thing since long. Thought I had to implement handling that protocol. Thanks a

Re: [OpenSIPS-Users] 2.2.x and CP with json

Ok, question part 2, now that the proper module is loaded :) From the MI interface in CP, I have 1 small issue. Using json to interface with opensips, I execute the command "*address_dump" *and execute. I receive the output of address dump | son:127.0.0.1:/json Successfully executed, no

[OpenSIPS-Users] [BLOG] Troubleshooting missing ACK in SIP

We all experienced calls getting self disconnected after 5-10 seconds - usually disconnected by the callee side via a BYE request - but a BYE which was not triggered by the party behind the phone, but by the SIP stack/layer itself.

Re: [OpenSIPS-Users] 2.2.x and CP with json

DOH! Got me. Sorry for the distraction but thanks for the simple fix. Working too fast on this new test bed I guess. Jeff Wilkie On Wed, Feb 22, 2017 at 11:20 AM, Bogdan-Andrei Iancu wrote: > Hi Jeff, > > Yes, you did overlook to load the dialplan module into your

Re: [OpenSIPS-Users] 2.2.x and CP with json

Hi Jeff, Yes, you did overlook to load the dialplan module into your OpenSIPS (the dp_reload command is prvided by this modules). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 02/22/2017 05:56 PM, Jeff Wilkie wrote: When using the mi_json

[OpenSIPS-Users] 2.2.x and CP with json

When using the mi_json module combined with CP, it appears that not all of the functions were ported over for this interaction to complete 100%. JSON enabled in opensips-cp/config/boxes.global.inc.php only allows some commands to work. One particular command that appears to not be working is

Re: [OpenSIPS-Users] Siptrace usage in 2.2.2

Hi Ionut, Am using the below config. Let me know if this is correct? But am not seeing any packets in the Homer side, rather seeing errors in Kamailio logs, which is mentioned below. Please help me to resolve this issue, if am doing anything wrong. Kamailio is running and listening on port 9060.

Re: [OpenSIPS-Users] Siptrace usage in 2.2.2

Hi, As regards *duplicate_uri* parameter it depends on whether or not you have *duplicate_with_hep* set. If *duplicate_with_hep *is not set (or set to 0) you should use*sip* else you should use *hep*. Regarding *traced_user_avp*, this parameter is obsolete and now moved as a parameter to

Re: [OpenSIPS-Users] rtpproxy not relaying data

Hi, Waldo! I only see the command for Update (initial request), I don't see the command for Lookup(200 OK). Moreover, are you sure RTP traffic gets to the rtpproxy machine? Because RTPProxy statistics doesn't see any packets getting to the server. Can you double check if the firewall allows

[OpenSIPS-Users] rtpproxy not relaying data

Hi: I am trying to use rtpproxy with my SIP proxy. It starts a session and both SIP clients send data to rtpproxy port as I see on wireshark after I modify the SDP part. I see data comming to rtpproxy but no data comes out of it. What could be going on wrong? Configuratoin issue? I send here

Re: [OpenSIPS-Users] ​MI fifo commands for multiple opensips instances

Thanks Aqs, for providing json module link. Yes we can have multiple options mi_datagram can also be used by running different instances on different datagram sockets. Thanks Robert, I am trying to deploy two different opensips instances using puppet on same server. I got your point we can