[asterisk-users] WebRTC and JsSIP
rnewton at digium.com
Wed Apr 16 16:03:42 CDT 2014
On Wed, Apr 16, 2014 at 1:35 PM, Consultor VOIP <voip at axys.com.br> wrote:
> Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.
> I configure my Asterisk 11.7.0 to work wit WEBRTC.
> Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at
> the Asterisk, but when we try to make a call they send a 488 response and
> finish it.
> here is the part of the SIP DEBUG
We can't do much with part of your debug. You'll want to post a
pastebin link to your full SIP trace, and be sure that it includes at
least VERBOSE messages turned up to 5.
Work on WebRTC support is on-going, so you'll want to test in the very
latest Asterisk version in your branch (11 or above). That means you
need to be on 11.9.0-rc2 at this moment.
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