14 Nov
2014
14 Nov
'14
12:51 a.m.
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(a)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
list
--
Sent from my mobile, and thus lacking in the refinement one might expect from a fully
fledged keyboard.
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
On 13 Nov 2014, at 22:38, Alex Balashov
<abalashov(a)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
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
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