On 7/9/2012 8:24 AM, Sergio Serrano wrote:
Hi all
I hope that someone of you can solve this. Right now I'm stuck!!!!!
I'm using asterisk with some SIP extensions. Basically I want to
establish a call between desktop voip phone (ext 181) and embedded sip
system (ext 182)
All I can see in CLI is:
== Using SIP RTP CoS mark 5
-- Executing [182@default:1] Dial("SIP/181-0000000a", "SIP/182")
in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/182
-- SIP/182-0000000b is ringing
-- SIP/182-0000000b is making progress passing it to SIP/181-0000000a
-- SIP/182-0000000b answered SIP/181-0000000a
-- Remotely bridging SIP/181-0000000a and SIP/182-0000000b
== Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-0000000a'
My guess is you need to add canreinvite=no to both SIP Peers in order to
avoid the re-invite which apparently is what is happening.
eRepublik - Join Me!
http://www.erepublik.com/en/referrer/csredes
Seems like extension 182 (called ext) is getting call and passing them
another time to me 181 (origin call)
I've try it with siemens pbx and works as expected
--
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