Re: [asterisk-users] Asterisk 11 WebSockets.
qasimakhan at gmail.com qasimakhan at gmail.com writes: 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-]: 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/16sip.conf: http://pastebin.pk/17http.conf: http://pastebin.pk/19extensions.conf: http://pastebin.pk/20Regards,Qasim -- _ According to the Asterisk developers, this is an issue in the hands of the browser developers. Here is the wiki page on the Asterisk 11 SIP over WebSockets: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support At this time, no media is flowing. James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 WebSockets.
Thanks :). Regards, Qasim On Wed, Sep 5, 2012 at 1:52 AM, James Mortensen james.morten...@a-cti.comwrote: qasimakhan at gmail.com qasimakhan at gmail.com writes: 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-]: 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/16sip.conf: http://pastebin.pk/17http.conf: http://pastebin.pk/19extensions.conf: http://pastebin.pk/20Regards,Qasim -- _ According to the Asterisk developers, this is an issue in the hands of the browser developers. Here is the wiki page on the Asterisk 11 SIP over WebSockets: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support At this time, no media is flowing. James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11 WebSockets.
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-]: *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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
