search for: newextens

Hi, I'm stucked in situation, and look for a work around if possible in Asterisk. I have a dialplan, [default] exten => 111222,n,Set(fu_callerid=141688xyxzz) exten => _X.,n,NoOp(Callerid ${fu_callerid}) exten => _X.,n,wait(2) exten => _X.,n,Answer() ? When, ?Answer Application is called AMI Event is triggered like this.. ? ? ? ? ? 'Event' => 'Newexten', ? ? ? ?

Friends, a new dialplan application is now available for testing : http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/ The corresponding feature request is located here : http://bugs.digium.com/view.php?id=12569 What can you do with it? Well, a direct usage of this application is to make an easy to use GoogleTalk voice gateway out of Asterisk. Here is an example (assuming the

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:3333/fetch,queuename) The idea is

Dear community members, I'm happy to announce that we now have code that allows you to use your XMPP (Jabber) client like a softphone to place SIP or PSTN (or whatever channel Asterisk supports) calls. The XMPP clients that support Jingle that I and others have tested are : - Pidgin (Linux, Ubuntu 9.10), version 2.6.2 : OK - Empathy (Linux, Ubuntu 9.10), version 2.28.1.1 : OK - Psi (Windows

I use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way to launch Dialplan function

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using

Hi all, just for learning purposes i made a little gui frontend that visualizes incoming and outgoing calls in realtime, using the events of asterisk. I experienced a strange behaviour for outgoing calls. The callerid for the *called* person got changed to one of my own numbers, after the channels git linked. After looking into the flow of events i saw that asterisk keeps sending an

On Thu, Feb 6, 2020 at 12:34 PM sduthil at wazo.io <sduthil at wazo.io> wrote: > On 1/29/20 2:31 PM, George Joseph wrote: > > For those of you who actually process SIP MESSAGE requests... Do you > > use any of the AMI events generated by the "Message/ast_msg_queue" > > channel? We want to change that channel to an "internal" channel that > >

I used to be able to pass dial strings to IAX2 providers with # characters at the end of the string. This is how we end dial strings for international calls. So, I would like to be able to selectivity chop off any # characters at the end of string, only if they exist. Basically as follows (chopping off the leading '9' with ${EXTEN:1} syntax: EXTEN from Phone EXTEN for Dial String

The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

; ; Display certain channel variables every time a channel-oriented ; event is emitted: ; ;channelvars =3D var1,var2,var3 So if you want fu_callerid, set: channelvars =3D fu_callerid And, once that variable is set, you should get a NewExten event, you should see the following key/value pair: ChanVariable(SIP/1234-00000001): fu_callerid=3Dfoobar --=20 Matthew Jordan Digium, Inc. | Engineering

<br> ;<br> ; Display certain channel variables every time a channel-oriented<br> ; event is emitted:<br> ;<br> ;channelvars =3D var1,var2,var3<br> <br> So if you want fu_callerid, set:<br> <br> channelvars =3D fu_callerid<br> <br> And, once that variable is set, you should get a NewExten event, you<br> should see the following

Hi all, what's your opinion about CDR-Event (like Hangup or Ring etc.) on AstManager ? Or, is something like this already implemented ? Regards, Thomas

Hi there, I am currently developing the ability to have a unified system/telephone login, with SIP phones paired to a computer. When a user logs into a computer, a notification is sent to an external service program which connects to Asterisk through the manager API. Besides that, the service program tracks user status on the computer, and triggers actions depending on various conditions.

Hey all, Something is wrong when i use originate command to call my phone (Asterisk1.4.22 + xp100 card). Actually, i have two problems. The first one: If i fire a outgoing call using originate command directly, after my pc startup, i will receive below error message: [Nov 26 07:58:53] NOTICE[6559]: channel.c:2898 __ast_request_and_dial: Unable to request channel Zap/1/13xxxxxxxxx but i can

hi i logged on to manager API from other terminal by telnet IPADDR 5038 now logged in with username mark let's say this connection Window A now i opened another connection with Manager API with same usename lets say this window B now if i give a command like originate,Redirect through window A connection , can i able to see its response:success/failure Originate:failed/succesfully