[asterisk-users] seems like call is picked and returned to me

[Andres]

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 at 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.

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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
>