<div dir="ltr">Hi Steve, <div><br></div><div>You mentioned that Kamailio was handling the registration of the clients...</div><div><br></div><div>Therefor, you want to do something like this if you want to be able to bridge calls:</div><br>exten => _900XX,1,Answer()<br> same => n,DumpChan()<br> same => n,Dial(SIP/${EXTEN}@<<kamailio-peer>>,30,t)<br> same => n,HangUp()<div><br></div><div>Another valid format:</div><div><br></div><div>exten => _900XX,1,Answer()<br> same => n,DumpChan()<br> same => n,Dial(SIP/<<kamailio-peer>>/${EXTEN},30,t)<br> same => n,HangUp()<br></div><div><br></div><div><br></div><div>Where "<<kamailio-peer>>" == the name of the peer you configured in sip.conf for Kamailio.</div><div><br></div><div>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).</div><div><br></div><div>Give a try and let me know how it goes.</div><div><br></div><div><br></div><div>Cheers, </div><div>Joel.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Dec 14, 2017 at 8:59 AM, Wilkins, Steve <span dir="ltr"><<a href="mailto:swwilkins@mitre.org" target="_blank">swwilkins@mitre.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p class="MsoNormal">Hello,<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Here is my extentions.conf<u></u><u></u></p>
<p class="MsoNormal">exten => _900XX,1,Answer()<u></u><u></u></p>
<p class="MsoNormal"> same => n,DumpChan()<u></u><u></u></p>
<p class="MsoNormal"> same => n,Dial(PJSIP/${EXTEN},30,t)<u></u><u></u></p>
<p class="MsoNormal"> same => n,HangUp()<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">I have 90001,90002 in pjsip.conf with a webrtc endpoint.<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Thank you,<u></u><u></u></p>
<p class="MsoNormal">-Steve<u></u><u></u></p>
<p class="MsoNormal"><a name="m_-7673493315878238017__MailEndCompose"><u></u> <u></u></a></p>
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<p class="MsoNormal"><b>From:</b> sr-users [mailto:<a href="mailto:sr-users-bounces@lists.kamailio.org" target="_blank">sr-users-bounces@<wbr>lists.kamailio.org</a>]
<b>On Behalf Of </b>Joel Serrano<br>
<b>Sent:</b> Thursday, December 14, 2017 9:29 AM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> Re: [SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio and Asterisk)<u></u><u></u></p><div><div class="h5">
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<p class="MsoNormal">Hi, can you share with us the asterisk dialplan part where you call the Dial() application?<u></u><u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">On Tue, Dec 12, 2017 at 06:38 Wilkins, Steve <<a href="mailto:swwilkins@mitre.org" target="_blank">swwilkins@mitre.org</a>> wrote:<u></u><u></u></p>
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<p class="MsoNormal">Hello All,<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">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.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">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.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">Thank you!<u></u><u></u></p>
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