[SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio and Asterisk)

[Wilkins, Steve]

Hello,

Here is my extentions.conf
exten => _900XX,1,Answer()
        same => n,DumpChan()
        same => n,Dial(PJSIP/${EXTEN},30,t)
        same => n,HangUp()

I have 90001,90002 in pjsip.conf with a webrtc endpoint.

Thank you,
-Steve

From: sr-users [mailto:sr-users-bounces at lists.kamailio.org] On Behalf Of Joel Serrano
Sent: Thursday, December 14, 2017 9:29 AM
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
Subject: Re: [SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio and Asterisk)

Hi, can you share with us the asterisk dialplan part where you call the Dial() application?



On Tue, Dec 12, 2017 at 06:38 Wilkins, Steve <swwilkins at mitre.org<mailto:swwilkins at mitre.org>> wrote:
Hello All,

I am looking for a Diagram or such that shows the flow of SIP traffic for a WebRTC Client1 => WebRTC Client2 call  using Kamailio in front of Asterisk.

I am unable to get Asterisk to find the correct registered clients, which are registered in Kamailio and am hoping verifying the flow will help give me a clue as to what is going on.  E.g. Using chrome and tryit-pjsip I have Client1, and Client2 registered in Kamailio. However when I try to connect Client1 to Client2 (make a call), Asterisk has no clue where Client1 and Cleint2 are registered to.

Thank you!
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