Displaying 20 results from an estimated 60000 matches similar to: "Spawn extension -----what does this mean ?"
2004 Sep 14
2
Spawn extension.....exited non-zero
I am recieving inbound calls to my asterisk box over IAX.
And getting "spawn extension....exited 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
2006 Apr 05
0
What does this error mean "app.c: Huh....? no dial for indications?"
Hi,
What does the following error mean:
Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications?
Here is the 'full' log around the error:
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to
agent '3002', on 'Local/510@default-6b6c,1'
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002
Apr 5 12:38:24 VERBOSE[22755]
2004 Nov 29
1
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
2003 Nov 05
2
spawn extension (inbound , h, 1) exited non-zero
2007 Sep 05
1
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
2005 Aug 20
1
Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension
Does VoicemailMan have to be installed ? Why not available. I have setup a
mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup
up using *97.
My *97 code in extensions.conf:
exten => *97,1,Answer
exten => *97,2,VoicemailMain(${CALLERIDNUM}@default)
exten => *97,3,Hangup
asterisk console:
Verbosity was 8 and is now 12
-- Executing
2004 May 05
1
strange sip behavior (looping back to my own extension vm)
Hello-
I am currently testing with a carrier that seems to be having some trouble
around toll-free (800 number) access. While a problem, its the resulting
behavior that I'm finding disconcerting.
When I dial an 800#, I get the following response:
-- Executing Macro("SIP/2700-e10b", "carrier-out|18005558355|70|r") in
new stack
-- Executing
2003 Apr 11
1
extension exited non-zero
I keep getting errors after every call hangs up:
*CLI> -- Starting simple switch on 'Zap/2-1'
-- Executing Macro("Zap/2-1", "roll2vm|Zap/9") in
new stack
-- Executing Dial("Zap/2-1", "Zap/9|18") in new
stack
-- Called 9
-- Zap/9-1 is ringing
-- Zap/9-1 is ringing
-- Zap/9-1 is ringing
-- Zap/9-1 answered Zap/2-1
2009 May 31
1
Problem releasing call from a SIP extension
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi all!
Making some changes in extensions.conf to test the incoming calls so that
these are derived to a SIP extension, I found something that draws attention
to me: if I test calling to my PSTN line from a mobile phone, when take the
call from the SIP extension (softphone), if the mobile phone releases the call,
sofphone do it too without problems,
2007 Apr 23
1
problem when using Dial(Local/extension@context)
hi folks,
I use Dial(Local/extension@context) to make calls received on my DID number
to ring a local extension. I notice that on 8 out of 10 calls, the audio is
NOT working in the incoming direction (DID provider to asterisk). Local
extension 2055 maps to SIP destination "homephone", and if i replace the
Dial(Local/2055@local) with Dial(SIP/homephone), it works fine 100% of the
2010 Aug 23
1
channel stay up when extension unreachable
Hi,
We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity
recorded in our full log. Could you help us to explain what had
happened. Thanks.
=== my friend, 801, from his room did a test by dialing echo test in
freepbx, *43:
[Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing
[*43 at from-internal:1] Answer("SIP/801-000003f5", "") in new stack
[Aug 20
2004 Jul 19
0
AGI Dial, Extension dial SIP Loop
At the moment I'm prototyping an advanced ENUM application with PHP
fetched from LDAP. When a user enters a full hostname as SIP adress I get
loop problems from the AGI EXECUTE DIAL and from a Dial in the
extension.conf.
-- Executing AGI("SIP/1000-c3c3", "enum.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/enum.php
enum.php: 123
enum.php:
2006 Jun 05
1
This should be easy: What happens when the Calling Party hangs up
svn trunk 31497
For the life of me, I can't get this :) I want to be able to catch the
situation where the calling party hangs up *before* the call is
connected to the called party. My dialplan is thus:
macro DialExternal(exten) {
Dial(Zap/G3/${exten},120,g,M(connected));
goto DialResult|r${HANGUPCAUSE}|1;
Hangup();
};
But the goto dialresult is not executed:
Executing
2005 Sep 06
1
one extension goes straight to voicemail, others don't
i have one extension going straight to voicemail, while others that are
configured identically don't, so i don't think it's an overall config
problem. nor do i think it's a callerID problem. maybe it's an enduser
operation that i can't find documentation on?
che*CLI>
-- Starting simple switch on 'Zap/48-1'
-- Executing Macro("Zap/48-1",
2006 Mar 23
0
Spawn-fcgi spawning & killing
While we were trying out a new capistrano setup tonight we noticed that
the spawner would alternately spawn fcgi processes, then the next time
it would kill them, then create, then kill, etc...
>From everything I can see/read/know, spawn-fcgi should not affect
running processes, right? Anyone else see this weird behavior? For now
we are using the ''check the port before
2008 Feb 09
1
Dialing SIP server user extension... Dial string issue...
Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial string but it's not working ...
Dial(SIP/gs102:test at 192.168.2.81);
User on sip server (192.168.2.81):
[gs102]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=gs102
secret=test
host=dynamic
dtmfmode=inband
defaultip=192.168.2.1
qualify=1000
mailbox=102
context=context-gs102
2012 Apr 04
1
npRmpi trouble - mpi.comm.spawn causes segfault
Dear all,
I have a large dataset of randomly generated weighed sample for which I
wish to compute a kernel density estimate.
I have used the "np" package successfully for smaller datasets, however
for the larger ones, they take too long when
using the cross validation options for bandwidth selection ("cv.ls" or
"cv.ml"). Of course, they are much quicker with
2005 Jul 21
0
Busy Extensions
Here is the output. These are Panasonic KX-TG2564's. Does something need to
be set for the phones? I can call out fine, but all of the extensions seem
to be busy.
Starting simple switch on 'Zap/5-1'
-- Executing Macro("Zap/5-1", "exten-vm|200@default|200") in new stack
-- Executing SetVar("Zap/5-1", "FROMCONTEXT=exten-vm") in new stack
2008 Sep 27
1
Problem with R on dual core under Linux - can not execute mpi.spawn.Rslaves()
Hi
I am trying to utilize my dual core processor (and later a
High-performance clusters (HPC) ) by using the Rmpi, snow, snowfall,
... packages, but I am struggling at the beginning, i.e. to initialise
the "cluster" on my dual core computer. Whenever I try to initialize
it (via sfInit(parallel=TRUE, cpus=2) or mpi.spawn.Rslaves(nslaves=2)
), I get an error message:
>
2007 Aug 02
1
A simple IVR extension problem
Hi list,
I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS
5.
I am having trouble to make my simple IVR extension work, here is relevant
config:
zapata.conf
----
context=incoming
signalling=fxs_ks
channel => 4
context=internal
signalling=fxo_ks
channel => 1
-----
extensions.conf:
----
[office]
exten => s,1,Dial(Zap/1,30)
[home]
exten =>