similar to: Manager redirect action does not appear to work in some cases.

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

Hi list, I experiencing a strange behaviour when transferring a call. The use case is like this: - Incoming call from Zap/1-1 - Routed to SIP phone SIP/1001 - The called user (SIP/1001) wants to redirect the call and presses "#" - IVR (default setup) says "Transfer" and user gets dial tone - User dials 1002 - IVR says "No such extension - please try again" ??? It

Hi, Any idears how to get call transfer to work? The "#" key is recognized but the following typed digits does not appear to be read and the IVR announce "Invald extension..." Debug output -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on Zap/1-1

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

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 > >

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

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

Hi. I am running asterisk 11 and i have usb 3g dongles to make my gsm calls with the following config in extensions.conf exten => _9X.,1,Dial(Dongle/dongle800/${EXTEN:1},120,KT) exten => _9X.,n,Hangup(${HANGUPCAUSE}) By dialing 9 it opens the dongle to make a call. I would like to restrict my caller id. so when i place a call from this dongle, it will send on the other end *blocked number*

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

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', ? ? ? ?

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 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

thanks a lot for the reply. i thought of that and i did try to send *exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)exten => _9X.,n,Hangup(${HANGUPCAUSE})* but the provider replies back that it is a wrong number. Then i inserted the sim to an ordinary mobile phone and dialed #31# and the number, then the call progressed fine and it restricted the number. What am i doing wrong

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

I did a yum update to my desktop machine as root this morning and now my regular logon account sees this whenever I press the enter key: etc/audisp/audispd.conf: Permission denied etc/audisp/plugins.d/af_unix.conf: Permission denied etc/audisp/plugins.d/syslog.conf: Permission denied etc/audit/audit.rules: Permission denied etc/audit/auditd.conf: Permission deniedetc/dhcp/dhclient.d/ntp.sh:

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

; ; 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