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

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On 13 Nov 2014, at 22:38, Alex Balashov abalashov@evaristesys.com

wrote:

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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

Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing

list

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sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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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