[SR-Users] Allow Re-invite mid call

[Alex Balashov]

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 at 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 at 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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--
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