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

[Joel Serrano]

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 at 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 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> 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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>
>
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