Re: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0
Returning from native bridge, channels: SIP/172.20.0.80-0819e0b8,
SIP/172.20.0.75:5090-081a35f8
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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>
> --
> /*/
> 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
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
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0
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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[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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[Asterisk-Users] Spawn extension -----what does this mean ?
for every call, * gives out : Spawn extension (default, 00x, 3) exited non-zero on 'SIP/201.50.117.161-081628e0' Spawn extension -what does this mean ? how to avoid this ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Spawn extension
--- Colin Anderson <[EMAIL PROTECTED]> wrote: Unfortunately, there doesn't seem to > be any kind of > granularity there so you can't branch based on what > went wrong, just that > something went wrong. hth. ok, but how can i fix this probleme. I don't know where to start with my trouble shooting. Unfortunately the SIP debug don't give more informations (only Service Unavailable). any suggestions?? Thx __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Spawn extension
> == Spawn extension (default, 998004, 3) exited >non-zero on 'SIP/2004-41dc' >What is the meaning of the exited non-zero line? afaik, after something executes, zero is returned: "Everything went OK" or -1 "Something bad happened", you can branch conditionally in the dialplan based on that. Unfortunately, there doesn't seem to be any kind of granularity there so you can't branch based on what went wrong, just that something went wrong. hth. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spawn extension
hi,
calling from Asterisk to PBX via Eicon Diva 4BRI gives
me the following error.
-- Executing NoOp("SIP/2004-41dc", " call for
998004") in new stack
-- Executing Dial("SIP/2004-41dc",
"CAPI/99:8004|20|r") in new stack
== Everyone is busy/congested at this time
-- Executing Congestion("SIP/2004-41dc", "") in new
stack
== Spawn extension (default, 998004, 3) exited
non-zero on 'SIP/2004-41dc'
What is the meaning of the exited non-zero line?
thx for your feedback
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Re: [Asterisk-Users] Spawn extension.....exited non-zero
On 14 Sep 2004 at 18:20, Matt Williams wrote:
> I am recieving inbound calls to my asterisk box over IAX.
> And getting "spawn extensionexited non-zero" errors.
> Im not entirely sure what this means, and would appreciate any help in
> fixing my problem. FYI: ** is the inbound phone number x.x.x.x
> is a remote asterisk box calling my own asterisk box.
>
> When I choose it to dial an internal extension using this dialplan:
>
> exten => *,1,Dial(${GRANDSTREAM},20);
>
> the phone "blips" and I get this error in the asterisk console:
>
> -- Accepting unauthenticated call from x.x.x.x, requested format = 8,
> actual format = 2
> -- Executing Dial("IAX2/[EMAIL PROTECTED]/1",
> "SIP/grandstream|20") in
> new stack
> -- Called grandstream
> == Spawn extension (inbound, *, 1) exited non-zero on
> 'IAX2/[EMAIL PROTECTED]/1'
> -- Hungup 'IAX2/[EMAIL PROTECTED]/1'
>
> And if I try to just answer the call and play a message using this
> dialplan:
>
> exten => *,1,Answer
> exten => *,2,BackGround(demo-congrats)
>
> I get this error:
>
> -- Accepting unauthenticated call from x.x.x.x, requested format =
> 8,
> actual format = 2
> -- Executing Answer("IAX2/[EMAIL PROTECTED]/5", "") in new stack
> -- Executing BackGround("IAX2/[EMAIL PROTECTED]/5",
> "demo-congrats")
> in new stack
> -- Playing 'demo-congrats' (language 'en')
> Sep 14 19:08:10 NOTICE[138140672]: chan_iax2.c:2375 iax2_read: I
> should never be called! Sep 14 19:08:10 WARNING[138140672]: file.c:548
> ast_readaudio_callback: Failed to write frame
> == Spawn extension (inbound, *, 2) exited non-zero on
> 'IAX2/[EMAIL PROTECTED]/5'
> -- Hungup 'IAX2/[EMAIL PROTECTED]/5'
>
>
> If this helps:
>
> I am running on FreeBSD (my previous asterisk server was also freebsd
> and with no problems) My processor is hyperthreading (a first for me -
> but i read there may be a problem with it?)
>
Yeah I have read there may be a problem with it to. If possible, it
would be worth testing with HT turned off and post your results.
Also, if this does not work, maybe it would pay to post some more of
your config files (i.e. obfuscated iax.conf, more of extensions.conf
and possible some of your sip.conf).
Cheers,
Matt Riddell
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[Asterisk-Users] Spawn extension.....exited non-zero
I am recieving inbound calls to my asterisk box over IAX.
And getting "spawn extensionexited non-zero" errors.
Im not entirely sure what this means, and would appreciate any help in
fixing my problem.
FYI:
** is the inbound phone number
x.x.x.x is a remote asterisk box calling my own asterisk box.
When I choose it to dial an internal extension using this dialplan:
exten => *,1,Dial(${GRANDSTREAM},20);
the phone "blips" and I get this error in the asterisk console:
-- Accepting unauthenticated call from x.x.x.x, requested format = 8, actual
format = 2
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "SIP/grandstream|20") in
new stack
-- Called grandstream
== Spawn extension (inbound, *, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]/1'
-- Hungup 'IAX2/[EMAIL PROTECTED]/1'
And if I try to just answer the call and play a message using this dialplan:
exten => *,1,Answer
exten => *,2,BackGround(demo-congrats)
I get this error:
-- Accepting unauthenticated call from x.x.x.x, requested format = 8,
actual format = 2
-- Executing Answer("IAX2/[EMAIL PROTECTED]/5", "") in new stack
-- Executing BackGround("IAX2/[EMAIL PROTECTED]/5", "demo-congrats")
in new stack
-- Playing 'demo-congrats' (language 'en')
Sep 14 19:08:10 NOTICE[138140672]: chan_iax2.c:2375 iax2_read: I should
never be called!
Sep 14 19:08:10 WARNING[138140672]: file.c:548 ast_readaudio_callback:
Failed to write frame
== Spawn extension (inbound, *, 2) exited non-zero on
'IAX2/[EMAIL PROTECTED]/5'
-- Hungup 'IAX2/[EMAIL PROTECTED]/5'
If this helps:
I am running on FreeBSD (my previous asterisk server was also freebsd and
with no problems)
My processor is hyperthreading (a first for me - but i read there may be a
problem with it?)
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Re: [Asterisk-Users] spawn extension (inbound , h, 1) exited non-zero
What does extension inbound, h, 1 have on it? is there an inbound,h,2? On Thu, 2003-11-06 at 08:51, Ali Mughrabi wrote: > Hi all , > > I'm facing the following problem , users are disconnected from > asterisk and I get the following message: > > spawn extension (inbound , h , 1) exited non-zero on 'Zap/xxx-x' > > and repeated for so many calls. > > have any body faced the same problem before? what might be the cause > and how can I sove it ? > > thanx in advance... > > > > __ > Tired of spam? Get advanced junk mail protection with MSN 8. > ___ Asterisk-Users mailing > list [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Enger <[EMAIL PROTECTED]> Xintegration ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spawn extension (inbound , h, 1) exited non-zero
Hi all , I'm facing the following problem , users are disconnected from asterisk and I get the following message: spawn extension (inbound , h , 1) exited non-zero on 'Zap/xxx-x' and repeated for so many calls. have any body faced the same problem before? what might be the cause and how can I sove it ? thanx in advance... Protect your PC - Click here for McAfee.com VirusScan Online ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spawn extension (inbound , h, 1) exited non-zero
Hi all , I'm facing the following problem , users are disconnected from asterisk and I get the following message: spawn extension (inbound , h , 1) exited non-zero on 'Zap/xxx-x' and repeated for so many calls. have any body faced the same problem before? what might be the cause and how can I sove it ? thanx in advance... Tired of spam? Get advanced junk mail protection with MSN 8. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
