similar to: Dialing out and recording

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

Hi, I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always over SIP) I want to keep track of who answered and of the length of the call. [outgoing-dev2] exten => h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished) exten => _X.,1,NoOp(Will send call to ${CC_DIALSTRING}) exten => _X.,n,Dial(${CC_DIALSTRING}, 60, M(uploadpeer-dev2^${CC_CALLID})em) exten =>

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

Hi, We have a system with both ISDN trunks and SIP. We receive incoming calls on both but always dial out via SIP. When dialing out the caller id is set like this: exten => _X.,1,Set(CALLERID(num)=${CC_ORIGNUM}) exten => _X.,n,Set(CALLERID(name)=${CC_ORIGNAME}) exten => _X.,n,Dial(${CC_DIALSTRING}, 60, em) This always works fine on SIP and on ISDN as well when the number is not hidden.

Hi, My problem is as follows. I registered an xlite client and dialed 1500 extension. In the extensions.conf i set as follows. exten=>1500,1,AGI(localhost// hello.agi. This hello.agi when connected plays a greeting message. Once this is connected from the script i want to transfer the call to another extension say 1600. How do i achieve this. I tried using ChannelRedirect but it didnt work.

Hi, i'm trying to implement the following scenario: - A user calls number 700 - Asterisk then dials to extensions 100, 200, 300, 400 and 500 - And then bridges all calls to a conference room I tried to use MeetMe and ChannelRedirect, but seems that after channel redirect nothing more is executed. So, this seem to work for the caller and first called, but the others

Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker "SIP peer authentication on an external database (RADIUS - LDAP)", etc. Although, none of these seems to give me the

Greetings folks; I have a bit of a conundrum, and I can't tell if Asterisk is doing something daft, or whether I'm clean missing out why it's doing what it's doing. So, I have a dialplan that looks a little like this: -------------------- [start] include => dids include => everythingelse [dids] ; Test exten => 8378,1,SetCallerID(3015551212) exten => 8378,2,Hangup

Hi List One more Problem I stumbled upon. Using Asterisk in a TSP environement. Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed. Example: +4198055615995555 +41 country prefix 98055 Routing Prefix 615995555 effective phone number Calls routed to Customers need to be put in the 'local' format. 0615995555 This is also the format of the From / To / Invite header

Hi all, I'd like to setup a really lean Asterisk installation that essentially has a full ISDN PRI (AT&T, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan) on a Digium TE207P adapter that all it does is convert the ISDN channels to SIP/IAX channels. Then I would add this Asterisk 'gateway' as a provider on one (or many) Asterisk systems on the back. With such a config I

Hi, I'm struggling with a feature in my home phone setup. I have several phones using both SIP and SCCP. What I try to do is to create a dynamic feature that works similar to the blindxfer feature built into Asterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Executing [123456 at NPDB2:76]

I receive calls over a T1 with callerid and then *ani*dnis*. I am able to strip out the ani and the dnis in the dialplan but when I try to set the caller ID to be the ani, it looks ok but then if I do a NoOp callerid on the next line, I get unknown. Here is the section of my dialplan: exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN}) exten =>

Guys, I am using Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux on 2013-01-18 19:52:25 UTC How can I set variable in one context and then Redirect a channel to another context and use variable there? The code below doesn't work, so I've got empty VAR1 in context_2 [context_1] exten => s,1,SET(__VAR1=VALUE1) exten =>

Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel

Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten => 125,1,Ringing exten => 125,2,Wait(3) exten => 125,3,Answer exten => 125,4,Wait(2) exten =>

Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf

I've noticed that with FastAGI (and maybe AGI) that when you sequentially send a sequence of dial commands, if the call is picked up, that after the call ends, the Fast AGI script keeps executing the commands! Is there anyway to stop execution once a call is picked up? I think looking at the result codes after the Dial to determine if the call was picked up or not is not a good idea... if it

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |