[asterisk-users] Asterisk 11 WebSockets.

qasimakhan at gmail.com qasimakhan at gmail.com
Mon Sep 3 06:25:15 CDT 2012


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

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