similar to: play sound on a specific channel

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

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/

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 =>

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

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

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

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

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

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

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

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

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

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 =>

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 {

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

Hi i have a Te205P connected to a PRI E1, can i force the outgoing callerid to change for each context? for example: [outgoing_context_one] ;force callerid to 12345 exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN}) [outgoing_context_two] ;force callerid to 22222 exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN}) Can i do that? thanks to all -- /*************/ nik600

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 /

Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMF&Channel=SIP/200-sdadsadkioah&Digit=1 the result is: <ajax-response> <response type='object' id='unknown'><generic response='Success' message='DTMF successfully queued' /></response>

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

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