Displaying 20 results from an estimated 2000 matches similar to: "rejected because extension not found"
2003 Nov 18
4
Help with Warnings
I'm trying to clean up some notices/warnings that are repeatedly logged
in *.Any Help would be appreciated as I'm not sure of the cause
/solution.
Here are the errors:
Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321
(zt_call): cidspill already exists??
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
/* Don't send audio while on hook, until the call
2009 Aug 26
1
app_swift issue
Hello
I have installed cepstral .... It works woderfull using an agi script but
.....
when i try to use Swift("say this") is Dial plan .... I get the error
[Aug 26 12:30:18] WARNING[7420]: pbx.c:3167 pbx_extension_helper: No
application 'Swift' for extension (actdemo, 123, 2)
Now i come to know to install app_swift
Here is the issue...
when i try to execute make command
2004 Jun 07
1
AVM B1 and PTP mode
Hi !
I've fetched a spare AVM B1 card from the cellar, and installed it. After
"modprobe b1pci" I did "capiinit" and capiinit moaned about a missing
t1.b4.
So I search the web and found one at
http://www.avm.de/ftp/cardware/b1/x_misc/ddi/. When I now look at the
controller, I finally see p2p-mode:
# cat /proc/capi/controllers/1
name b1pciv4-a400
io
2009 Aug 26
4
Fw: app_swift issue
Hi Shakeel,
I had the same problem building app_swift (1.6..) myself and searched the web far-and-wide for a solution. I eventually contacted Darren Sessions -- who was maintaining that plug-in -- about a month ago. He was involved in another project and said he might be able get to it after a few weeks. But, since then, his website http://www.darrensessions.com/ has gone out of comission.
I
2003 Mar 03
0
Voicemail Volume Control Patch
Hello all,
This is my first attempt at posting a patch.
So if I screw this all up, my apologies and please
someone let me know without beating me up too bad.
To use this patch. You need to have an extra line
in /etc/asterisk/voicemail.conf that looks like
volgain=10.0
The 10.0 gets passed to sox which you will need installed
on your system. 10.0 is what works for me, anything over 1.0
will
2006 May 16
2
Multiple Registers
List,
Does anyone know how to limit the amount of registrations that a sip user
can have?
For example, I have 2 softphones that I use on my laptop & desktop, both use
the same username & password. If I have both softphones up at the same time,
I can make simultaneous calls with each of them.
I know you can have call-limit=1 but in this case, I want to allow them to
have 3 way calling
2003 Jun 04
1
new application Dialtone()
Hello,
I created a new application for myself called Dialtone() by modifing
res/res_indications.c file. It can be used as such:
exten => s,4,Dialtone(30|${CALLERIDNUM})
exten => s,5,Playback(time-exceeded)
exten => s,6,Goto(s|1)
It will stutter if you have new voicemail and you have passed the mailbox
number as I did above. It will stop dialtone the moment you press a key
2014 Mar 13
1
Any help Address 0xfffffffe out of bounds in app_confbridge.casterisk-11.5.1 using confbridge.conf
Address 0xfffffffe out of bounds why and how to
solve.MyConfbridgeCount(conferencenumber,variablename )return total number
of user in conference given by conferencenumber otherwise zero.At runtime
using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call
function count_exec(struct ast_channel *chan, const char *data).But at
compile time char * data cause core dumped.
2003 Jul 03
1
res parking patch
Ok, a little patch that adds a little functionality to call parking.
With that, you can pickup the older parked call, if many are in the
parking lot. The default exten to do that is 750, but can be changed
by setting "parkpick => exten" on parking.conf , like
[general]
parkext => 800 ; What ext. to dial to park
parkpos => 801-820 ; What extensions to
2009 Oct 05
3
Questions about app_jack.c
Hello,
My configuration is :
Card 0 - kernel dummy sound card
Card 1 - my soundcard
I have a jackd running in background. My jackd launch command is :
jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
--capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
--outchannels 2 --dither triangular &
1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to
2003 Jul 08
1
RTP.C codec error 19
hi ..
when placing a SIP call to a sip host in the states
every few seconds I get an RTP codec 19 error.
I know this is related to comfort noise, and the
call goes through OK ... how can I suppress
the error message ?
Also, many times I get "Invalid CSeq Number"
back from 216.52.153.207 (which is the host
i'm calling) and the call drops.. is there a solution
for this ?
cheers
Dave
2003 Jul 11
4
module : cdr_sybase.so
If anyone is interested ... just in case! :-)... I have tried to write ,
based on the cdr_mysql.so module, an Sybase module.
To compile you can use something like that:
export SYBPLATFORM=linux
export SYBASE=/opt/sybase
cc -I$SYBASE/include -c -o cdr_sybase.o cdr_sybase.c
cc -shared -Xlinker -x -o cdr_sybase.so cdr_sybase.o -lsybdb -lm
-L$SYBASE/lib
(anyone could write the corect Makefile
2003 Jul 22
2
enabling dtmf detection on zap channel?
Hi,
is there a way to enable dtmf detection on zap channels? I am trying to
pickup, play a ringtone and the dial out. I.e.
exten => s,1,Wait,1
exten => s,1,Answer
exten => s,2,Playtones(dial)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => _X,1,StopPlaytones
exten => _X,2,Dial,Zap/g8/BYEXTENSION|10
2001 Jul 05
1
OpenSSH Logging Madness
Feature request:
- Please add a new LogLevel corresponding to the LOG_NOTICE syslog level.
- Then modify OpenSSH to log to LOG_NOTICE only these events:
- login failures
- login successes
Specifically, please:
- add a new element to the LogLevel enum, say, 'SYSLOG_LEVEL_NOTICE',
between 'SYSLOG_LEVEL_INFO' and 'SYSLOG_LEVEL_ERROR', in log.h
-
2005 Aug 25
2
Custom Application For Asterisk
Hi All
I just completed a custom application for Asterisk (i
m not a C guru so i just copy codes from other
application and alter according to my needs)
attached files is the source file
this application is working fine but still i need you
people to give suggestion to improve it
Primary task of this application is to get a parameter
from extensions.conf, query sql server and play a
files
2010 Sep 22
1
T38 and codecs negotiation
Hi,
I'm working with asterisk 1.4.35 and found an issue regarding codecs
negotiation when T38 is enabled (t38pt_udptl=yes).
In particular if the INVITE sdp contains no allowed codec the call is not
rejected with "488 - Not acceptable here" but it goes through and the 200 OK
SDP is as follows:
v=0
o=root 27285 27285 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
2016 Aug 05
2
Toll free pattern matching
I have this in my config:
exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/tollfree/1${EXTEN})
exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/tollfree/${EXTEN})
exten => _NXXNXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/trunk/1${EXTEN})
exten =>
2009 Oct 22
2
carefulwrite: write() returned error: Broken pipe
Dear,
I am getting this in CLI on release candidate version of Asterisk. Any
ideas, or points where to look?
-- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi
[Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write()
returned error: Broken pipe
-- <SIP/916-fc001968>AGI Script rad-auth.agi completed, returning 0
Best regards,
Josip
2010 Oct 12
0
rtpip patch
Hello *,
is the rtpip patch still valid for asterisk 1.6 (with some code
changes, obviously)?
https://issues.asterisk.org/view.php?id=8161
Or, in asterisk 1.6 there is an alternative to using it?
This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11
--- chan_sip.c 2010-10-12 13:47:49.000000000 +0200
+++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200
@@ -987,9 +987,6 @@
2014 Mar 12
0
module load Crash Asterisk 11.5.1 app_confbridge.c
=====================================================================
Asterisk-11.5.1 Centos6 app_confbrige.c
=====================================================================
APP: MyConfbridgeCount(Confbridgename,variablename)
it will return no of user in conference if conference is created or else
zero.
Task: Using Dailplan user want to retrive no of user in conference
'6050'