Hi Steve, 

You mentioned that Kamailio was handling the registration of the clients...

Therefor, you want to do something like this if you want to be able to bridge calls:

exten => _900XX,1,Answer()
        same => n,DumpChan()
        same => n,Dial(SIP/${EXTEN}@<<kamailio-peer>>,30,t)
        same => n,HangUp()

Another valid format:

exten => _900XX,1,Answer()
        same => n,DumpChan()
        same => n,Dial(SIP/<<kamailio-peer>>/${EXTEN},30,t)
        same => n,HangUp()


Where "<<kamailio-peer>>" == the name of the peer you configured in sip.conf for Kamailio.

At the end, the key here is that you are calling SIP/XXXX which is not local, if you call PJSIP/XXXX (per your example) you are trying to reach a local endpoint (in this case, the device that would have to be directly registered with Asterisk).

Give a try and let me know how it goes.


Cheers, 
Joel.

On Thu, Dec 14, 2017 at 8:59 AM, Wilkins, Steve <swwilkins@mitre.org> wrote:

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@lists.kamailio.org] On Behalf Of Joel Serrano
Sent: Thursday, December 14, 2017 9:29 AM
To: Kamailio (SER) - Users Mailing List <sr-users@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@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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