VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
voip at axys.com.br Guest
|
Posted: Wed Apr 16, 2014 1:36 pm Post subject: [asterisk-users] WebRTC and JsSIP |
|
|
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
<--- SIP read from WS:177.64.122.237:49217 --->
BYE sip:500@187.122.82.197:0;transport=ws SIP/2.0
Via: SIP/2.0/WS e8ilhkrhlup2.invalid;branch=z9hG4bK7306188
Max-Forwards: 69
To: <sip:500@177.64.122.237>;tag=as52a1a298
From: "G" <sip:8000@177.64.122.237>;tag=ue84kn6rku
Call-ID: u5hkiispkvn9g841oede
CSeq: 9338 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0
Some one can help me with this problem?
Thanks
Gerald |
|
Back to top |
|
|
rnewton at digium.com Guest
|
Posted: Wed Apr 16, 2014 4:04 pm Post subject: [asterisk-users] WebRTC and JsSIP |
|
|
On Wed, Apr 16, 2014 at 1:35 PM, Consultor VOIP <voip@axys.com.br> wrote:
Quote: | 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
--
_____________________________________________________________________
-- 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 |
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|