similar to: Originate (Executing a System Command)

Hey Gang, Trying to figure out how I can do the following (have each part working individually but drawing a blank on combining) 1) click on-screen which sends an AMI originate (works fine) 2) the originated call is to an internal extension that looks up the number to be dialed (works) 3) then via Perl, adding in a SIPAddHeader for answer-after=0.. (works separate from the above) What I

When using call files I see Dial command when on the CLI however when using the AMI I dont see the originate Dial. Is there a logging level that will show the originate dial on the CLI? If there isnt currently a log state for that shouldnt it do that just like the call files do? My logger.conf has: ; debug => debug console => notice,warning,error,dtmf ;console =>

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center Anahi Ludue?a escribi?: > Hi, thanks Miguel. > I have another question: if I want to call the GoSub

Hi, I would have thought that when spawning a channel using the Originate() dialplan command, variables prefixed with two underscores would be preserved. However this does not work in the following case. Dialplan code: [intern] exten => 200,1,Set(__myvar="foo") exten => 200,n,Originate(Local/123 at test_orig,exten,dummy) [test_orig] exten => 123,1,NoOp(${myvar}) exten =>

Date: Fri, 1 Oct 2010 18:40:40 -0300 From: Rodrigo Lang <rodrigoferreiralang at gmail.com> Subject: Re: [asterisk-users] AMI Originate To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <AANLkTikV+32vKVSkAFmkDciOPn+rO=k3jYJmsZLNj1QS at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 3

Hi All, I'm trying to use the orginate cmd. I have it working if originate is from a user e.g. SIP/7777 originate SIP/7777 extension 987654321 at outbound-route What i'd like to be able to is instead of a local extensions i would call an outside number then connect it another outside number. e.g. originate SIP/85431210 at outbound-route extension 987654321 at outboudn-route is this

Does anyone know of a way to make an originate request coming over the management interface (e.g. AstTapi click-to-dial) include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer? I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg,

Hello all, I am quite new in asterisk and I am trying to create a dialplan that executes the following steps: 1. A SIP friend dials 102 extension. 2. Asterisk PBX responds with some beeps. 3. The sip friend hangs up the phone. 4. Asterisk PBX calls back to the sip friend after 30 seconds with the application music on hold. I tried to implement this using h extension but I got the following

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi, With Asterisk 1.8.3, I can't figure out how to pass an argument to a macro which is used within an originate command. Here is my sample dialplan to illustrate: exten => 123,1,Answer() exten => 123,n,Originate(SIP/20,app,Macro,foo,bar) exten => 123,n,NoOp(This is the NoOp after the originate command) exten => 123,n,Wait(30) exten => 123,n,Hangup() [macro-foo] exten =>

Hello! I?ve an issue whit CDR using asterisk 1.4.23.1. I?ve configured mysql to store cdr information, but, while I put into cdr_mysql.conf the field ?userfield=1? and doing a query I found that this field is empty in the cdr table. On the other hand I can?t find records in the cdr table that show me calls generated through AMI using Originate Action, that?s calls are not stored in the CDR, but I

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to keep the DND status of my Snom phones and the astdb in line but I'm stuck on integrating my gui DND button which talks to * using the manager interface (actually it uses Astmanproxy as the gui host is on a different network to asterisk and can't see the Snom's across the network). All's working fine in my Dialplan; when someone dials the code for DND-on or

Hi all, When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is no 'ActionID' to tie it back to the command. How do you guys deal with this? -------------- next part -------------- An HTML

Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I

Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager???? Thanks and regards

>>>>> Jon Skoien <jon.skoien at jrc.ec.europa.eu> >>>>> on Wed, 7 Dec 2016 11:04:04 +0100 writes: > I would like to ask once more if this is reproducible also for others? > If yes, should I submit it as a bug-report? > Best, > Jon Please Windows users .. this is possibly only for you! Note that I do *not* see problems on

Dear All, I am originating the call directly to the SIP Provider using the maganger interface + originate (ASYNC) command. Here is the PHP-AGI Script. $call = $asm->send_request('Originate', array('Channel'=>"SIP/416XXXXXXX at ABC/n", 'Context'=>'ORIG',