similar to: Set up two PSTN calls and then join them

Displaying 20 results from an estimated 200 matches similar to: "Set up two PSTN calls and then join them"

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 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:
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
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: $oSocket =
2007 Jul 08
1
Asterisk Help
Hi I need help in configuring a auto dialer system using Asterisk. I'm holding my customers number in MySQL want to fetch 10 numbers one time and dial if gets connected and answered by customer wants to play a sequence of message . Please help . I've tried here is my code to place calls but in this I see no of failure calls are more than 50%. so please advise.
2011 Feb 13
1
Call Files, Variable passing
Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($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,
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 Oct 27
0
EM.One
Pat Phelan posted this on Facebook - thought the SIP functionality would interest some people here as well. Jajah inks large Japanese dealJajah inks large Japanese dealJajah inks large Japanese deal (Pat Phelan <http://www.facebook.com/profile.php?id=754567533> ) <http://www.facebook.com/profile.php?id=754567533> By Pat Phelan
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
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,
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
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 03
0
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
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
2007 Jul 12
0
No subject
don't have a public facing web page but you are looking for people to click on but a personalized list of numbers. In order for someone to access this directory you are going to be asking for a username/password correct? If so just tie the username to a selection of 'my location' checkboxes that I tick and then the app remembers this location next time I log in (eg server side
2008 Sep 23
3
Fwd: more on Free World Dialup groups and FWDLive
FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I think it will fail but we shall see. I really don't get the nerve of them (Free World Dialup has changed it's name to FWD) to ask for free ideas and development on a non-free service. Maybe if they can come up with a killer app and people will adopt it, then it might work, but then again, people
2009 Jul 19
1
Re: skype
What phone works with Skype when the computer is turned off? I am looking around for a phone to use with Skype. I Really don't want to spent 140 bucks for one. Can you use any VOIP phone or does it have to be skype? Also i wanted to know what phones I can use to were I don't have to have my pc on 24/7.
2007 Oct 13
0
swfdec patch, adds letterSpacing support
Hello, here is a first small patch that adds support to Flash 8 letterSpacing property as described here : http://wiki.mediabox.fr/documentation/flash/textformat/letterspacing http://livedocs.adobe.com/flash/9.0_fr/main/wwhelp/wwhimpl/common/html/wwhelp.htm?context=LiveDocs_Parts_bak&file=00002274.html