My guess, then, is that the reinvite lacks appropriate attributes of an in-dialog message. Otherwise, it should be getting routed normally in the loose_route() section.
On 13 November 2014 17:49:24 GMT-05:00, Eric Koome ekoome@yahoo.com wrote:
Using stock 4.x configuration available at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
On 13 Nov 2014, at 22:38, Alex Balashov abalashov@evaristesys.com
wrote:
On 11/13/2014 05:36 PM, Eric Koome wrote:
How do I process re-invite from sip provider mid call which check connection. At the moment Kamailio replies 404 not here, and call is dropped.
Can you post your Kamailio config?
-- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States
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Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com