similar to: Problem: features (from features.conf) not available if call was originated by manager API or call file

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

My company for various reasons has asked that I come up with a way to have previously parked calls be re-parked in the same parking slot. I have looked at setting up asterisk so that the receptionist chooses which slot to place a call, but I think there is an easier way. That is when I came up with the idea of "Sticky Park". Here is how it would work. A call would come in and

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hello all, I want to park calls using the callpark application, but I don't want to hear the saydigit when the called is parked. To resolve this issue I use the following instruction in the dialplan: exten => _8XX,1,ParkAndAnnounce(|1000|local/10 at default|) Because local/10 at default is not defined to a peer I get a lot of warnings. :( Is there a better way to resolve this

Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing ? follow = true 5. Endwhile 6. UnParkCall 7. Go on .. The idea is let the call waiting while I do some things on the dialplan, is it possible?? Maybe is not parking the solution?? Thanks

Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website

Just upgraded to 1.8, we use the multi lot parking feature by dialling *4. We are not getting the parking slot announcement being played to the person who parks the call, so it's impossible to tell which slot they've gone into. Could someone check our config? On Debian Squeeze using packages from http://packages.asterisk.org/debsqueeze main (Asterisk 1.8.11.1-1digium1~squeeze)

I have a SIP phone calling via a SIP trunk another asterisk system, that then sends the call out a ZAP channel. When I press any of the features defined in features.conf, The end user on the ZAP side hears the DTMF tones, and none of the features work. My DTMFmode on the SIP users definition is rfc2833 Asterisk console doesn't register that a feature is being recognized, any ideas? Below

Hi, Been playing with Call parking, and I can`t help but wonder if I am doing something incorrectly. The way I understand it (using default config in features.conf), is I would transfer a call to extension 700, which would park the call, tell me "701". I could then hang up, go fetch the fright person and tell him "call 701 you have a call waiting for you". The way I

I am trying to use call parking. I have the following in features.conf [general] parkext => 700 parkpos => 701-720 context => parkedcalls When I try #700 from my softphone asterisk just passes it and doesn't interpret it. Can someone tell me what I am missing? I am using asterisk-1.2.17 Thanks, Alberto

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,

> -----Original Message----- > From: Ronald [mailto:asterisk107@gmail.com] > Sent: 26 May 2005 10:47 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Little Php question > > > Hi > I'm trying to make a call from a local webpagee through my > xlite softphone > (xlite1) > BTW when I'm trying to do it through

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:

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

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,

Hi I think you should have a look at the end of line - you are missing " :-) Br, dmirty -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Ronald Sendt: 26 May 2005 11:47 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Little Php question Hi I'm trying to make

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.

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");

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

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,