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

[Olle E. Johansson]

9 jul 2012 kl. 15:24 skrev Sergio Serrano:

> 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'
> 
> 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
> 

It's very hard to see what's happening without seeing the SIP logs. You just see that something went wrong in the process of setting up the bridge.

/O