Hi, I was testing with newly introduced websocket support in asterisk 11. I have successfully implemented everything except when i try to make a call i get no audio. I have tried both SipML5 as well as SIP-JS as clients. the call get connected but i never hear any audio stream. I however get the following warning
WARNING[2626][C-00000000]: *chan_sip.c:9686 process_sdp:* Ignoring video > stream offer because port number is zero > When i turn rtp debug on i can see RTP getting through. *CLI Output*: http://pastebin.pk/16 *sip.conf*: http://pastebin.pk/17 *http.conf*: http://pastebin.pk/19 *extensions.conf*: http://pastebin.pk/20 Regards, Qasim
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