[asterisk-users] WebRTC and JsSIP

Rusty Newton 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.[1]

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[2] at this moment.

[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
[2]: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.9.0-rc2.tar.gz

Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

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