Hi Sergio Thanks for the reply. There was a problem in my ffmpeg (livavcodec) which was not buit with videocodec support.I have replaced it and now not getting the error. But a strange problem I am facing now. I have tried transcoding between h263 and h263+.I have used Xlite and linphone. I am calling from linphone which is using h263-1998 codec; App transcoder encodes the incoming h263-1998 to h263 and places call to xlite. It is also evident from the sip signalling traces that codec between asterisk and linphone is h263-1998 and between asterisk and xlite is h263.But if i configure xlite only for h263 ; no video is apperaing. But if i keep codec in xlite h263-1998 (i.e h263+) video appears. I am not sure if app_transcode module is really encoding in h263 format thogh log says it is encoding.
Thanks and regards Anand On 02/11/2009, Sergio Garcia Murillo <[email protected]> wrote: > > Hi anandadip > > Get the core dump and a back trace of asterisk when it seg faults > > Best regards > Sergio > > anandadip mandal escribió: > > Hi > I want to make video call between two sip phone having different video > codecs using app_transcoder. > I have used the following dialplan > [default] > exten => 101,1,Answer > exten => 101,2,transcode(,1...@default,h...@qcif > /fps=12/kb=52/qmin=4/qmax=12/gs=50) > exten => 102,1,Dial(SIP/101) > > the 102 ( having h263-1998 codec) extension is calling 101 (having h263 > codec). > I can see the call between the two phone established but no video; also i > dont see any ack coming from 101 and within seconds asterisk gives a > segfault. > Without app transcoder, video call works fine when both phone use h263-1998 > codec. > I am using asterisk 1.4; the transcode module loads succesfully; even it > executes and places a call to the configured extension) > > Please help me if i am using the correct dialplan or am i missing > something. > > Any help will be much appreciated. > > Regards > Anand > > > > On 26/10/2009, anandadip mandal <[email protected]> wrote: >> >> Hi >> I have successfully compiled and able to load the app_transcoder.so; >> I want to know the configuration of extension.conf to put the >> app_transcoder in use. >> I have two sip soft phone(video capable) 3000, 3001 which are already >> registered to asterisk and I can make audio call between them; >> Also please let me know if i have to add anything specific to >> extesion.conf and sip.conf for enabling video call. >> Any help will be very much appreciated. >> Thanks and regards >> Anand >> >> >> 2009/10/20 anandadip mandal <[email protected]> >> >>> is there any document for compilation procedure of app transcoder?also >>> could someone point me how to integrate it with asterisk? >>> Thanks >>> Anand >>> >>> >> >> >> -- >> Anandadip Mandal >> > > > > -- > Anandadip Mandal > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-video > > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-video > -- Anandadip Mandal
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