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

2007-09-12 Thread nik600 
-081a35f8
Sep 12 12:33:13 DEBUG[3648]: chan_sip.c:2450 sip_hangup:
update_call_counter(caller) - decrement call limit counter
Sep 12 12:33:13 DEBUG[3648]: app_dial.c:1661 dial_exec_full: Exiting
with DIALSTATUS=ANSWER.
  == Spawn extension (default, 1002, 2) exited non-zero on
'SIP/172.20.0.80-0819e0b8'
-- Stopped music on hold on SIP/172.20.0.80-0819e0b8


On 9/6/07, nik600 [EMAIL PROTECTED] wrote:
 yes, i've tried asterisk -r

 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 [EMAIL PROTECTED] wrote:
  Have you tried asterisk -rvvv?
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] 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/[EMAIL PROTECTED])
 
  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/[EMAIL PROTECTED]) in new 
  stack
  -- Called [EMAIL PROTECTED]
  -- 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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 https://sourceforge.net/projects/ccmanager
 https://sourceforge.net/projects/nikstresser



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https://sourceforge.net/projects/nikstresser

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

2007-09-05 Thread nik600 
Hi

i generate a call from the dialplan in this mode:

exten = 1002,1,Answer()
exten = 1002,2,Dial(SIP/[EMAIL PROTECTED])

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/[EMAIL PROTECTED]) in new 
stack
-- Called [EMAIL PROTECTED]
-- 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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Re: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0

2007-09-05 Thread Shonga_Kerz 
Have you tried asterisk -rvvv?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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/[EMAIL PROTECTED])

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/[EMAIL PROTECTED]) in new 
stack
-- Called [EMAIL PROTECTED]
-- 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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Re: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0

2007-09-05 Thread nik600 
yes, i've tried asterisk -r

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 [EMAIL PROTECTED] wrote:
 Have you tried asterisk -rvvv?

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] 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/[EMAIL PROTECTED])

 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/[EMAIL PROTECTED]) in new 
 stack
 -- Called [EMAIL PROTECTED]
 -- 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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 http://mail.yahoo.com


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https://sourceforge.net/projects/nikstresser

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