similar to: Need help with perl script/agi for ringback

Displaying 20 results from an estimated 200 matches similar to: "Need help with perl script/agi for ringback"

2013 Oct 10
2
utils.c: fwrite() returned error: Broken pipe how to solve it ???
Dear all, I want to make call through socket i have set code given below: #!/usr/bin/perl -w use IO::Socket::INET; sub asterisk_command () { # my $command=$_[0]; my $ami=IO::Socket::INET->new(PeerAddr=>'127.0.0.1',PeerPort=>5038,Proto=>'tcp') or die "failed to connect to AMI!"; print $ami "Action: Login\r\nUsername:
2006 Jan 10
2
Problem with Action:Originate with ASterisk Manager
Hi Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel:
2005 Feb 28
1
Manager "Message: Originate failed" beinggenerated when callee does not pick up
<<I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.>> Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf
2003 Oct 13
1
PRI/E1: machine freeze/dies after a few calls
Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1
2011 Jan 10
0
No subject
Wait a second... Action: DBGet\r\nFamily: DS\r\nKey: 0733025975\r\n\r\n In the dialplan: exten =3D> 0106024975,1,Set(DB(DS/0733025975)=3DINUSE) exten =3D> 0106024975,n,Hangup() exten =3D> 0106024976,1,Set(DB(DS/0733025975)=3DUNAVAILABLE) exten =3D> 0106024976,n,Hangup() Just a short call to my cell phone, to se if i get anything back, my = cell phone doesn=E2=80=99t even ring. Wait
2011 May 19
2
click to call with php
Hello, i have asterisk 1.4 installed and i want to use click to call in order to do an outbound call if there is any php code in order to do this operation thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110519/417ac394/attachment.htm>
2005 Jan 20
2
API Call Bridge?
Hi All, Does anyone know of a way to dial two different outbound numbers and bridge them together using the Asterisk API? Cheers, Taff. --------------------------------- ALL-NEW Yahoo! Messenger - all new features - even more fun! -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 Sep 14
1
Open file for reading and writing with APPEND
Hi, I want to keep a running file of some statistics generated by a running process. Occasionally the process will be restarted. On restart, I'd like to read the last line of the log file to set some variables to their last state, and then continue to append values to that same file. I can't seem to get the appending part to work. I can read values form the file, but then upon the
2005 Sep 08
1
How to increase delay before incoming call answer with tdm400p
Is there a way of increasing the delay before asterisk picks up the incoming PSTN call? I'm using a tdm400p with fxo card. It seems to pick up the inbound call immediately. I want to delay detecting the call by about 10 secs if poss. Done some searching but couldn't find anything relevant. Cheers, Taff!!! ___________________________________________________________ Yahoo!
2004 Jun 22
2
Any echo issues with phones from TDM400P > X100P
Hi, I'm thinking of purchasing a TDM400P card and was wondering if anyone has experienced any echo issues with phones off these cards connecting to the PSTN via the X100P cards? I have had my fingers burnt with a Voip phone > X100P. Cheers, Taff. --------------------------------- ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself -------------- next part
2011 Apr 05
1
allpage issu on asterisk 1.8.3.x
Hey Guys! I have perl script for allpage which is working fine with asterisk 1.8.2.3 version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is there anything changes ? If i run this script from command like it works but not from asterisk dialplan. This script nothing but just connecting AMI interface and using Variable: SIPADDHEADER=Alert-Info: Ring Answer variable to
2009 Sep 27
0
Is channel local what I need?
On 1.6.0.16-rc1: I'm using app_fax.so to send a fax, and then send a confirm. 'send' => 1. Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config] 2. System(env echo -e "Channel:DAHDI/g0/........\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >${UniqueFile}) [pbx_config] [ Context 'fax-tx' created by
2004 Nov 22
8
Patching asterisk for spandsp
When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this.
2009 Mar 30
3
Call-limit=1 breaks attended transfer
Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff..
2009 Sep 27
1
digium fax: failed to queue document
In my quest to actually send a fax, I'm now stuck trying to send the confirm. First I send the fax: -- Executing [send at outbound-fax:2] System("Console/dsp", "env echo -e "Channel:DAHDI/g0/12036378447\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >/var/spool/asterisk/outgoing/call-1254012878.0") in new stack -- Auto fallthrough, channel
2004 Jun 27
3
Re:Latest Echo changes
Hi, I've tried the latest CVS Zaptel and Asterisk after following the Echo fix threads. But echo is the same if not worst. Has anyone managed to alleviate their echo from these latest changes? --------------------------------- ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 Mar 26
3
Checking status of a cell phone
Hi, I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. Below example is for a ?Swedish? cell phone, dont know if it works in the same way for other countries. I could define ?redirecting? numbers for 3 traffic cases when u dial my mobile (073-302 59 75): NOT_INUSE call forward to A INUSE
2013 Jan 29
1
Sweave files generating miktex errors
Dear useRs-- I have been using Sweave with miktex for years, but on a new install on Windows XP, miktex seems to be hung up on single quotes. See example below. Digging through stackexchange, I found using \usepackage[noae]{Sweave} in the tex file solved the problem. My questions are: --Why would this happen? I have the ae package installed. --If inserting [noae] is the solution, how do I
2013 Feb 20
2
exten => h,n,AGI(generateCall.php,${NEXT})
not able to run my php from AGIi am using asterisk 1.8.13 (debian)i am able to make call file using php command line..but when executing php from AGI, it is not working..kindly see the attachment if bellow text is not readable...___________________________________________________ File: /etc/asterisk/extensions.conf[call]exten => call,1,Answerexten => call,n,Playback(hello-world)exten =>
2004 Jun 21
0
A Callback AGI script
Hi there, I just give you the script (in Python) I have just written in case of someone would like to implemant this. I think it is more simple than the one we can see over the net... It uses DISA (security issues ==> limit access with contexts and the password !!) and CAPI but it should work with type of channel. Basically, you ring your asterisk and the line goes down after 1 ring. Asterisk