[SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

Pan Christensen pan.christensen at phonect.no
Fri May 11 16:19:56 CEST 2018

Hello Steve.

What are you trying to achieve?

The call could go from client A to Kamailio to client B. No need to involve Asterisk. If you need PBX functionality, the INVITE needs to be routed to Asterisk, which will most likely answer the call and then set up a new call to client B. As Asterisk doesn't know where client B is, it needs to route this new call to Kamailio where client B is registered. It's possible for Asterisk to know where client B is but that solves nothing and may create other problems.

With kind regards
Pan B. Christensen
Phonect AS

From: sr-users <sr-users-bounces at lists.kamailio.org> On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
Subject: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

Hello All,

I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using Kamailio and Asterisk.

Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back to Kamailio.  Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered in Kamailio for the called number.
Does this seem correct?  It seems like there is an extra hop in there.

The reason I am now very curious now is because everything works fine if using Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.

I would appreciate any thoughts on this.

Thank you in advance.

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