[OpenSIPS-Users] rtpproxy and record calls

2016-12-27 Thread Denis via Users
Hello! I try to use rtpproxy for call recording and have two problems 1) rtpproxy records only one way of the call (from callee to caller). For starting rtp proxy i use rtpproxy_engage("conrf",,"1",) function. 2) i am using top_hiding with "C" flags (which should change callid) and i noticed, that

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

2016-12-27 Thread Denis via Users
Hello, Flavio! Thank you very much for your help! I made some test and it worked.But, in additional, i want to ask you about g729 codec. In rtpproxy documentation says that beginning from 2.0 g729 codec supported.In the dictionary of extractaudo i see some g729* files. Is there necessary to

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

2016-12-27 Thread Flavio Goncalves
Hi, Yes you can extract audio from rtpproxy. The extractaudio utility is very handy and you can compile with G.729 from the linphone project bcg729. It is very easy to use, simply use the utility followed by the name of the recording without any extension. Check the source code for the other

Re: [OpenSIPS-Users] b2b server_address parameter

2016-12-27 Thread Răzvan Crainea
Hi, Ziv! Are you using any advertise line in your script? If you do, you should add it for the listener the message was received on, i.e. listener=udp:PRIVATE_IP:5060 as PUBLIC_IP Note that this will change the IP for all the headers, including Via, Record-Route, etc. If you don't want to

Re: [OpenSIPS-Users] custom presence bodies

2016-12-27 Thread Răzvan Crainea
Hi, Tito! Yes, the Dialog id is mandatory. You can find more info in Section 4.1.1 of RFC4235[1]. Now I am not sure what you are trying to do, but if you use the pua_dialoginfo module[2], OpenSIPS will be able to generate the Publish for each call. [1]