[asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0

[nik600]

yes, i've tried asterisk -rvvvvvvvvvvvvvvvv

i've also tried sip debug, but i can't reach any error... only that
the cmmunication is finished.

On 9/6/07, Shonga_Kerz <shonga_kerz at yahoo.ca> wrote:
> Have you tried asterisk -rvvv?
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of nik600
> Sent: Wednesday, September 05, 2007 9:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero
> on 'SIP/host-0819d0d0
>
> Hi
>
> i generate a call from the dialplan in this mode:
>
> exten => 1002,1,Answer()
> exten => 1002,2,Dial(SIP/user at host)
>
> the call is generated, but after some seconds it is interrupted, here
> the asterisk log:
>
> *CLI>     -- Executing Answer("SIP/host1-0819d0d0", "") in new stack
>     -- Executing Dial("SIP/host1-0819d0d0", "SIP/caller at host") in new stack
>     -- Called caller at host
>     -- SIP/host-081a2610 is ringing
>     -- SIP/host-081a2610 answered SIP/host1-0819d0d0
>     -- Attempting native bridge of SIP/host1-0819d0d0 and SIP/host-081a2610
>   == Spawn extension (default, 1002, 2) exited non-zero on
> 'SIP/host-0819d0d0'
>
> i've enabled sip debug, but nothing interesing has been showed
>
> host1 is an SJphone and host is a software that implements SIP protocol.
>
> Can you help me to guess where is the problem?
>
> if i try to create a call from SJphone 2 SJphone all works fine.
>
> Is possible that exists a problem in asterisk ?
> where ? how can i find it ?
>
> thanks to all
>
> --
> /*************/
> nik600
> https://sourceforge.net/projects/ccmanager
> https://sourceforge.net/projects/nikstresser
>
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-- 
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser