Thanks :).
Regards, Qasim On Wed, Sep 5, 2012 at 1:52 AM, James Mortensen <james.morten...@a-cti.com>wrote: > 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-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/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
