similar to: Problem sending a DTMF remotely. Please need help...

Displaying 20 results from an estimated 1000 matches similar to: "Problem sending a DTMF remotely. Please need help..."

2009 Oct 02
0
Sending a DTMF remotely with PlayDTMF problem.
Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer
2009 Oct 06
0
Problem sending a DTMF remotely. Please need help!!!
Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended
2009 Oct 05
1
Problem sending a DTMF remotely. Please need help!!
Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer
2008 Jul 07
1
DTMF on iax channel is not interpreted by asterisk
Hi! I'm using asterisk 1.4.17 with twinkle and a custom phone based on iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf. While the twinkle client is able to initiate an attended transfer using *2 (as configured in features.conf), the iax client is not. I can see the DTMF messages showing up on the asterisk console, but asterisk does not invoke the features
2004 Aug 12
1
AgentLogin issue
Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --
2004 Aug 24
2
Voicepulse incoming / dial extension
All: I am trying to use Voicepulse as my incoming line and want the caller to simply dial the extension of the party they want to reach. Here is my problem: - the first time they dial it works fine and I see the following on my console Aug 24 23:14:31 DEBUG[-1126876240]: chan_sip.c:4408 build_route: build_route: Contact hop: <sip:6035057098@66.234.228.137> --
2011 Jan 12
1
DTMF not being heard correctly by far end conference system
Hi there I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some
2011 Feb 13
1
Call Files, Variable passing
Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket,
2012 Dec 12
1
Asterisk 11 originate errors
Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1
2007 Jul 06
1
Asterisk Manager
Hi this is my code for * manager: $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die("Connection to host failed"); fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret:
2013 Feb 23
0
click2call with AMI?
Hi, I have a PHP code with AMI to using in click2call system. here is my code: $user = "usernamr"; $secret = "secret"; $channel = 'SIP/' . $sip; $context = "from-internal"; $waitTime = "20"; $timeout = 20000; $priority = "1"; $maxRetry = "2"; $pos = strpos($number,
2007 Jul 08
1
Asterisk Help
Hi I need help in configuring a auto dialer system using Asterisk. I'm holding my customers number in MySQL want to fetch 10 numbers one time and dial if gets connected and answered by customer wants to play a sequence of message . Please help . I've tried here is my code to place calls but in this I see no of failure calls are more than 50%. so please advise.
2009 Dec 23
1
AMI originate and PHP
Hi Guys, I am trying to make a web form where a person is allowed to put in $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller ID. There are a few problems that I am facing with Asterisk AMI Originate command. The reason why I want to use the darn AMI Originate is because I am sending calls to mobile phones and I want to have some accountability and to know if a call was
2009 Dec 18
2
To Asterisk AMI Gurus - Tacking issue with originate
Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax
2007 Oct 13
0
Set up two PSTN calls and then join them
I wish to set up two PSTN calls and then connect them similar to Jajah (is this called 3pcc?). The PSTN interconnect is handled by a third party SIP provider. I can do this using the manager or call files. An example (using php) would be: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n");
2006 Dec 29
0
PHP to call script
Using the php script below. I am able to enter my number and the number to call, however I get the following error: -- AGI Script cid-spoof.agi completed, returning 0 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 -- Executing Wait("OutgoingSpoolFailed",
2007 Jul 08
1
Early Media Handling
Hi using php script and Asterisk manager I'm dialing numbers and once gets connected send to an exten in my dial plan that plays an automated message but some time without answering even it goes to my exten. How can I handle early media in Asterisk that is I want only when user answer the call it should goto my specified extension. my php script: $oSocket =
2007 May 05
2
Manager API Output
Hi, Is there any way that I can store my manager API output that is: My question is that is there any why using that I can get the QueueStatus and store the result in some text file for further processing. <?php $strHost = "127.0.0.1"; $strUser = "cron"; $strSecret = "1234";
2007 Jan 02
5
Call connected, cannot hear or speak - $20 for fix
I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one
2005 Sep 13
1
wctdm, issue w/outbound calls
Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked