On 7/9/2012 8:24 AM, Sergio Serrano wrote:
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.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'
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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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