Displaying 20 results from an estimated 500 matches similar to: "What would cause a high memory usage in pbx_spool.c ?"
2009 Feb 03
2
some kind of timeout problem in pbx_spool.c
I am using outgoing call files. I typically see the "ooh something
changed / timeout" on a regular bases every second to be exact.
Then it stops until some other call event happens.
So I "mv" my call file to the outgoing spool directory, I am listening
to that message, another call file is "mv"'ed into the directory
and something happens to the timeout that its
2006 Jun 06
0
pbx_spool - outgoing qcall failure upon call progress
Does anybody have a work around for this problem.
I create a call file in /var/spool/asterisk/outgoing and Asterisk picks
it up and starts placing the call.
However if the called channel provides any sort of progress indication
(such as a SIP or IAX channel indicating ringing that causes the console
to say "SIP/xxxx is ringing") the code in pbx_spool.c indicates a call
failure and
2003 Jul 17
0
Example: Writing a click-to-call application using pbx_spool
I have written a small perl CGI script that demonstrates how one might
use the asterisk spooler 'pbx_spool' to make a 'click-to-dial' type
application. The script is intended to be a demonstration example only
and since it has little security, should not be deployed. I was just
experimenting with the spooler and wrote this to try some things, and I
though it'd be a good example
2004 Jul 29
6
Zaptel doesn't see remote hangup ? euro-isdn
Hi
Just received my spanky new TE405P today to replace my Cisco gateway...
After much fiddling (I forgot to switch it to E1) I got it to work and
everything "seems" to work perfectly on our ISDN PRI.
If I dial-in from the PSTN to a SIP phone, the call goes through and if I
hangup either the SIP phone or the remote end, the call gets disconnected
and destroyed
However, if I dial-in
2003 Oct 08
2
pbx_spool and contexts
When I drop my file into the outgoing folder, the call is completed but
the 'Context' entry is not respected. Instead, it drops into the default
context. It does drop "properly" into the default context and function as
would be expected. I looked through the source but didn't see any reason
it would be completely ignoring the context.
Call file: (where
2005 Jul 16
2
Memory leak in asterisk CVS
Hi,
My Asterisk CVS is apparently not doing much (other than keeping SIP &
IAX2 registrations alive and doing some ZAP calls (without
echo-cancellation), but slowly the memory is filling up, so much so that
100m virtual memory is used up within 12 hours and I have to restart the
asterisk application every 48 hours to make sure I have enough memory...
How can I help resolve this problem?
2004 Jul 15
3
G.729 codec doesn't seem to work *even* after installing the license
Hi,
I am trying to post this again as I am getting no answers and the
support@digium.com bounces...
(I have searched the whole list and can't find the answer either)
I have installed a 5 user license for G.729 and want to route calls through
Asterisk from my G.729 phone to Cisco AS5300 also using G729.
Both Cisco and the phone connect using this codec if I do not force the call
to go
2014 May 15
1
Call file problem, DelayedRetry/retrying spite MaxRetries: 0
I am using Realtime extensions as well, in case that would matter.
Following problem arises from time to time, a call will successfully
terminate:
[May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing
[t at project_init:1] Hangup("SIP/peer-2-00002f7e", "")
[May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init,
t, 1) exited non-zero on
2011 Jun 15
1
call file challenge...
Greetings!!
We're getting some strange results using call files.. no matter the
technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason
(3) Remote end Ringing" message when attempting to originate a call from a
call file. Numbers changed to protect the innocent....
using call file....
//------------CALL FILE------------//
Channel: DAHDI/g1/918005551212
2009 Mar 11
4
Are .call files working with extensions.ael ?
Hello,
With an extensions.ael enabled system, I keep getting whatever I change into
my "astup.call" file :
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least
one of app or extension (or keyword message/pdu) must be specified, along
with tech and dest in file /var/spool/asterisk/outgoing/astup.call
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:
2013 Jun 24
2
Asterisk-11 loop behaviour
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4.0
FreePBX = 2.11.0.2
Snom870 Handsets
We are in the process of moving to an Asterisk based PBX. At the
moment most things work as we wish. However, I have just notices that
when I force a reload using 'amportal a reload' I see this loop start
in 'asterisk -rvvvvvvvvvv':
> Channel Local/s at tc-maint-000002a4;1
2006 Mar 06
3
call manager integration
I am getting this error from call manager (4.0) and asterisk 1.2.4
I have canreinvite=yes on the call manager setup.
I can call into the asterisk box from call manager. THat seems to work.
When I am calling out of the box using a call file I see
this entry from call manager...
What might be the problem with my setup?
THanks,
JErry
----------------
<Date>03/06/2006
2003 Jul 23
1
Newbie Help
Hi - after hearing others rave about * I thought I'd have a go - extract
from a 'make' on a stock debian system as follows... (I tried to post the
whole make up to this point but it was too big for the list)
make[1]: Leaving directory `/usr/src/asterisk/channels'
make[1]: Entering directory `/usr/src/asterisk/pbx'
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
2015 Dec 31
2
Logging to CDR after call file Not Answered
If the end user does not pick up the phone, is there a way to log to the
CDR about the call file failing?
/var/log/asterisk/messages does show a NOTICE message
[2015-12-31 06:58:46] NOTICE[28059] pbx_spool.c: Call failed to go through,
reason (3) Remote end Ringing
[2015-12-31 06:58:46] NOTICE[28059] pbx_spool.c: Queued call to SIP/102
expired without completion after 0 attempts
but I would
2006 May 24
1
Placing call files in /var/spool/asterisk/outgoing/ does not work
Hello everyone
I'm trying to make asterisk get a call out using the .call system.
The setup is A@H 2.6
This is the content of the file is :
<<<
Channel: Zap/g0/052MYPHONE
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: ext-local
Extension: 210
Priority: 1
>>>
I'm
2007 May 22
1
Local SMS how-to.
Hello,
i just want to activate SMS service between my asterisk local sip accounts
and between asterisk and local sip accounts. How can i do this thin? Also i
tried smsq to an account but all i obtained is a error message:
---<Cut Here>---
May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open
/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission
denied, deleting
May 22
2005 Mar 21
1
DTMF doesn't seem to get through incoming ZAP channels
Hi,
I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium
TE410P card.
Calling into meeting rooms that have been configured with the p option
works fine.
From ZAP extensions the # key does not work to exit, however from SIP
extensions the # key works fine. This makes me believe that somehow the
DTMF doesn't get through the ZAP interface. After furter experimenting
2004 Jul 29
1
Re: Zaptel doesn't see remote hangup ?
Thanks Peter,
Yes, indeed the problem seems to be exactly what you describe. It's overhere
the same. If I dial a mobile number it disconnects immediately when I hangup
the mobile. But for analog numbers it takes around 10 seconds or so...
Well, at least now I know how to debug pri :-)
Walter.
On Thu, 29 Jul 2004, Walter Klomp wrote:
> However, if I dial-in from the SIP phone to my
2009 Oct 09
1
${REASON} not getting set.
Hi all,
I've got a program that creates a callfile and most if it working great.
However, when a call fails, I'm trying to capture the reason, which I'm told
should be in the ${REASON} channel variable.
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Here is an excerpt from the callfile:
Channel: local/155555555
Callerid:Tests <155555555>
MaxRetries: 0
RetryTime:
2007 Apr 24
2
Funky BIND/named errors
I have been getting these for awhile now in my log files.
Apr 24 11:02:38 asterisk named[1072]:
lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53
Apr 24 11:02:38 asterisk named[1072]:
lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53
Apr 24 11:02:38 asterisk named[1072]:
lame server resolving 'pbx_spool.so' (in