[asterisk-users] WebRTC and JsSIP

Wed, 16 Apr 2014 11:36:13 -0700

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:[email protected]:0;transport=ws SIP/2.0
Via: SIP/2.0/WS e8ilhkrhlup2.invalid;branch=z9hG4bK7306188
Max-Forwards: 69
To: <sip:[email protected]>;tag=as52a1a298
From: "G" <sip:[email protected]>;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 






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