Hello Pan,

Thank you for responding!,

In my configuration I have to use Asterisk as my PBX, and I use Kamailio in front of Asterisk accepting and inspecting calls.

I have many AORs, for which a phone can register to.  I have noticed that, depending on the call, when a call arrives it gets REGISTERED in

Kamailio (I do not forward the registration to Asterisk); Kamailio sends the INVITE to Asterisk, and then Asterisk send the INVITE back to

Kamailio.  The calls seem to work fine (Duplex Audi/Video).  I think I know why the INVITE is forwarded to Kamailio but have not been able

to work around it; the AOR’s all have a Contact of Kamailio and I think this is the reason for the forwarding of the INVITE.  If I don’t set Kamailio as the Contact for the AOR’s, calls do not work.

Thanks Again,

-Steve

From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Pan Christensen
Sent: Friday, May 11, 2018 10:20 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

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
Developer
Phonect AS 

From: sr-users <sr-users-bounces@lists.kamailio.org> On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List <sr-users@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 WebRTCóWebRTC 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.