Displaying 20 results from an estimated 600 matches similar to: "force outgoinc callerid"
2007 Apr 21
3
FAX on PRI and TE205P
Hi
i have a PRI connected to a TE205P.
Actually, can i send and receive FAX through Asterisk using stable solutions?
Or shall i connect an ATA to Asterisk and then a modem with Hylafax?
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2008 Jun 14
1
play sound on a specific channel
Hi to all
can i play a sound or a dtmf tone on a specific channel using AMI?
Thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2007 May 07
2
h323 problem with asterisk 1.2.18
i am experiencing problem with asterisk 1.2.18
I've downloaded and installed pwlib and openh323 with the following commands:
cd /path/to/pwlib
./configure
make clean opt
cd /path/to/openh323
./configure
make clean opt
then 'ive set the corresponding PATH
PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/
export PWLIBDIR
OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/
2007 Oct 27
0
Call center manager for Asterisk (Release 0.5)
CCMANAGER 0.5 released!!
NOTE:
this is a previous alpha release, maybe there is some customization to
do on the settings files,
i can't write a clear and complete howto at the moment
I don't have released upgrades in the last months but the project is still alive
i'm too busy at the moment, i'm following other projects to have some
resources (both money and time)
and then i can
2007 Aug 03
2
partial ChanSpy
Hi
is it possible to spy (not record, spy) partially on a channel?
for exaple, i'd like to listen only the input or the output voice.
is it possible?
thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 Jun 29
4
asterisk call unique id in dialplan
Hi
how can i retrieve the call unique id of asterisk in the dialplan?
I have enabled the cdr logging on a postgres database.
In the table cdr each record has a field that assumes an unique id
(for example: 1141628669.51)
Can i retrieve this from the dialplan?
For example:
exten => 203,1,Answer
exten => 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id})
exten =>
2008 Jul 17
1
OpenH323 and ptlib version for asterisk 1.4.21.1
Hi what version of openh323 and pwlib are suggested for asterisk
1.4.21.1.? Thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
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
2007 Jun 19
2
PhpAgi call generation
hi
i'd like to write a simply application in php with phpAgi that:
- connect to Asterisk
- call an external number using a Zap channel
- play a message
here is some code:
<?php
$asm = new AGI_AsteriskManager();
if ($asm->connect()) {
$asm->Originate("Zap/g1/1","number","default","1");
/*
play message...
*/
} else {
2007 Nov 20
1
store 2 separate records in cdr when a call is transferd
Hi
i've read this post
http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html
I just want to know if there are some upgrades... on 1.4 or 1.2.
I'd like to store two records in the CDR instead of one, when a call
is transferd.
Is it possibile now?
Thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
2008 Jan 08
2
disable call waiting by default
I've connected some analogic phone to some fxs modules on an analogic card.
I want to disable by default the call waiting sound.
I know that dialing *70 before to call the call waiting is disabled
until the next call, but isn't there a setting or a dialplan command
to set up this automatically?
Thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
2007 Dec 23
3
OpenVox A800P01 and ZT_CHANCONFIG failed
Hi
i've got an openvox a800p01 with 1 FXO and 4 FSX
i've done the following:
- downloaded zaptel-1.4.7.1
> >> - downloaded the file opvxa1200.c
> >> - copied in zaptel-1.4.7.1/
> >> - edited makefile adding opvxa1200 in the modules and the voice
> >> opvxa1200.o : zaptel.h wctdm.h
> >> - edited zaptel.sysconfig adding
MODULES="$MODULES
2007 Dec 11
0
new Asterisk installation with openvox 1.2 or 1.4?
Hi
i need to install a server with this hardware:
1 OpenVox B800P
1 OpenVox A800P01
4 OpenVox FXS-100 FXS100
4 OctWare SoftEcho SOFTECHO
Do you suggest 1.2 or 1.4 branch?
Is now 1.4 stable ?
I've tried 1.4 the last year but i've experienced many problems with
misdn drivers.
Thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
2008 Jul 09
0
disable DTMF on a particular channel
Hi to all
is it possibile (via AMI or dialplan) to disable the DTMF tone on a
particular channel?
Thanks in advance
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2008 Jun 04
0
Patch for app_asr.c: DTMF instead of goto
Hi to all
if someone of you is interested on it, i've changed the code of app_asr.c
With these patch you can use the ASR application to play DTMF tones,
so you can have your own AGI application that uses the ASR and manages
the DTMF tones without change the dialplan.
EXAMPLE
exten => 003,1,Ringing
exten => 003,2,Wait(3)
exten => 003,3,Answer
exten =>
2007 Sep 13
0
[phpAGI] generate a call from a SIP device to Asterisk
Hi
i need to generate a call from a SIP hardware device to Asterisk.
The device isn't registered with a sip account to Asterisk.
What i've done, is to do this (using phpAGI):
.....
$asm->Originate(SIP/user_on_device at ip_of_device,2000,"default","1");
.....
And on the extension 2000 in the context "default"
exten => 2000,1,ChanSpy(|g(100))
exten
2007 Oct 03
0
multiple iax users on the same host
Hi
i'm setting up a hylafax server, using iaxmodem to talk with asterisk
(asterisk and hylafax are both on the same lan).
Can i setup on the same host (Hylafax) multiple iax accounts ? (each
account is used by a iaxmodem instance).
The account can be on the same port or should i change the port for
each iax account?
Thanks
--
/*************/
nik600
2007 Dec 15
0
OpenVox B800P and asterisk 1.4/ mISDN-1_1_7
Hi
i've installed this software:
******************** SOFTWARE
mISDN-1_1_7
mISDNuser-1_1_7
Asterisk-1.4.15
******************** SOFTWARE
misdn is correctly loaded by misdn-inist start
Here there is the misdn.conf (copied from an existing and working
installation with Asterisk 1.2.x and one BN8S0)
******************** MISDN.CONF
[general]
misdn_init=/etc/misdn-init.conf
debug=0
bridging=no
2007 Mar 14
3
Call center manager for Asterisk (Release 0.3)
Hi
i just want to let you know that is available a new release of ccmanager.
I've added the possibility to import queue_log information in a mysql
database and to generate reports using this information.
The software is in a beta state and provides this functionality:
- users management
- call generation (making a GET or POST request on a certain URL)
- queue management (LOGIN / LOGOUT /
2006 Feb 11
2
configure TE205P on asterisk@home
hi
i'm trying to configure a TE205P on asterisk@home
i've edited /etc/sysconfig/zaptel adding this line:
MODULES="$MODULES wct2xxp"
now, when the system is loading, i can see that the wct2xxp module is
loaded correctly
but if i try the command:
/usr/local/sbin/genzaptelconf
i get:
STOPPING ASTERISK
STOPPING FOP SERVER
Generating '/etc/zaptel.conf'
Generating