Hi Alex,

On a WebRTC to WebRTC call (both UAC's registered in Kamailio) =>

- When Asterisk sends the INVITE back to Kamailio, it has the id of the call 
and the IP Address of the laptop the call was made from 
INVITE 
sip:[email protected];transport=ws;alias=133.148.203.97~53778~6;alias=133.148.203.97~53778~6;ob
 SIP/2.0

- I have been using IP Addresses instead of Domains, as I have had much more 
luck doing it that way.

Should I switch to FQDN and Domains?  If so, may I ask what all would need to 
be changed?

Thank you for your time. 

-Steve

-----Original Message-----
From: sr-users [mailto:[email protected]] On Behalf Of Alex 
Balashov
Sent: Sunday, May 13, 2018 10:32 AM
To: Kamailio (SER) - Users Mailing List <[email protected]>
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of 
an "INVITE"

That seems correct and sounds like it should work. What is the request URI of 
the INVITE that comes back into Kamailio on the B leg? Does its domain match 
the AOR domain? Do you have your registrar/usrloc set to use domains on 
lookup()?


-- Alex

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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