|
From: VEYSSIERE N. <nic...@th...> - 2014-10-10 07:40:42
|
Thanks for your help about the endpoints, in fact I wasn’t very proud of how I did the multi-port thing (I didn’t noticed the listener thing, my bad). I will change it right away. But this doesn’t really answer my question : for my problem I am instantiating only 1 endpoint for sip and 1 for h323 (+ the local), I am able to receive call, but calling a remote party doesn’t works as describe earlier. I manage to enable logging, and I found this strange trace : ………………………….. call.cxx(486) Call GetMediaFormats for Call[……]-EP<sip>[…………………..] EMPTY LINE HERE So there seems to be a media format problem, and here is what I don’t understand : I am able to receive call from any format that opal is able to decode, but when I want to generate a call no media formats are enabled ? This is certainly a miss from me, but I don’t know where. Thanks. -- Nicolas V. De : rob...@gm... [mailto:rob...@gm...] De la part de Robert Jongbloed Envoyé : vendredi 10 octobre 2014 01:39 À : VEYSSIERE Nicolas Cc : opa...@li... Objet : Re: [Opalvoip-user] MakeConnection initiate a call, but no rtp packets are generated Why are you using more than one endpoint for each protocol? You can have as many "listeners" on as many ports as you like on each endpoint. There is no need for "1 port / endpoint" as you describe. That is really not the way the system is designed, so you are probably on your own. You are also very likely to be broken in some future version. I do not recommend that path. Robert Jongbloed OPAL/OpenH323/PTLib Architect and Co-founder. On 9 October 2014 20:31, VEYSSIERE Nicolas <nic...@th...<mailto:nic...@th...>> wrote: Hello, I am using opal as a record system, so far so good, every calls (sip or h323) are recorded as wave without problems. I have now to code the fact that my program must initiate the call. Because I have a lot of endpoints (more than 10 for h323 and more than 10 for sip, because of 1 port / endpoints), I don’t use setupcall in my manager, but I am using MakeConnection directly in the endpoint. The call seems to works, as openphone that I am trying to call is receiving the call, but there is 2 problems : - Openphone stats for tx / rx audio are stuck at 0ko, and no rtp packets are received (using wireshark) - 32s after the beginning, the call is cleared automatically from my side. Can someone help my find where is the problem ? Thanks, best regards. -- Nicolas V. ------------------------------------------------------------------------------ Meet PCI DSS 3.0 Compliance Requirements with EventLog Analyzer Achieve PCI DSS 3.0 Compliant Status with Out-of-the-box PCI DSS Reports Are you Audit-Ready for PCI DSS 3.0 Compliance? Download White paper Comply to PCI DSS 3.0 Requirement 10 and 11.5 with EventLog Analyzer http://pubads.g.doubleclick.net/gampad/clk?id=154622311&iu=/4140/ostg.clktrk _______________________________________________ Opalvoip-user mailing list Opa...@li...<mailto:Opa...@li...> https://lists.sourceforge.net/lists/listinfo/opalvoip-user |
