asterisk users - Sep 2007 - 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

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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