On Fri, Sep 4, 2015 at 7:51 PM, <asandoval...@gmail.com> wrote: > Hello everyone. I'd appreciate a lot your help with this issue. I'm running > a very basic script of JS for subscribing my jsSIP User Agent to my local > Asterisk server and making a voice call. I don't get any warnings or errors > from the Asterisk CLI, but when I make a call to a legacy SIP phone or SIP > trunk well configured, there is no audio on any side although there is > ringing, calls can be answered and they never drop. > > The IP address of the SIP messages is correct both in the header of the > message and in the RTP description, and it succeeds with sending ICE > candidates. My Asterisk 12 was compiled with SRTP and pjproject. I don't get > any error or warning messages on Asterisk, and I suppose that the SIP > messages are ok. > > I read at the Asterisk WebRTC Wiki > (https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support) this: > "Starting with Asterisk 12 you need to have pjproject libraries installed, > otherwise you most likely won't have audio in your WebRTC calls and no > warning whatsoever!" > I properly installed it and selected it for the Asterisk compilation, but I > wonder wether I did it wrong, and how can I check it ... > > These are my files: > > http.conf > [general] > enabled=yes; > bindaddr=0.0.0.0; > bindport=8088; > prefix=asterisk; > tlsenable=yes; > tlsbindaddr=0.0.0.0:8089; > tlscertfile=/etc/asterisk/keys/asterisk.pem; > tlsprivatekey=/etc/asterisk/keys/asterisk.pem; > > rtp.conf > [general] > rtpstart=10000; > rtpend=20000; > icesupport=true; > stunaddr=stun.l.google.com:19302; > > sip.conf > [general] > context=toSipTrunk > allow=ulaw > allow=alaw > allow=gsm > > [1000] ;legacy softphone (zoiper) > secret=****** > type=friend > host=dynamic > dtmfmode=rfc2833 > disallow=all > allow=ulaw > allow=alaw > context=myContext > > [1001] ;jsSIP User Agent > type=friend > username=1001 > host=dynamic > secret=****** > encryption=yes > avpf=yes > icesupport=yes > directmedia=no > transport=udp,ws > force_avp=yes > dtlsenable=yes > dtlsverify=no > disallow=all > allow=ilbc > allow=g729 > allow=gsm > allow=g723 > allow=ulaw > dtlscertfile=/etc/asterisk/keys/asterisk.pem > dtlsprivatekey=/etc/asterisk/keys/asterisk.pem > dtlssetup=actpass > context=myContext > > ... Thanks in advance
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