Thank you for the response. This is the first time that I require help on these topics and I thought this list was the correct one. I've already sent my subscription request to the asterisk-user mailing list, with the hope that I find the answer to my problem soon.
Regards On Mon, Sep 7, 2015 at 1:15 PM, Matthew Jordan <[email protected]> wrote: > On Fri, Sep 4, 2015 at 7:51 PM, <[email protected]> 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 > > The asterisk-dev mailing list is for discussions regarding the actual > source code of Asterisk. Please use the asterisk-users mailing list > [1] for deployment, setup, troubleshooting, and other related > questions. > > [1] http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Matthew Jordan > Digium, Inc. | Director of Technology > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev >
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