similar to: About outdail SIPCALLID

Actually my problem is not related to sip.conf and extensions.conf. I have used only standard files from martin pdf which are given as example. I am able to call some system connected over LAN, when each has a softphone and are registered on a asterisk server. But now what i want is instead of using the softphone I write a function in my file which will be executed when the call is placed. In that

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

Hey all, I've been fighting with this all morning, and I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and

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();

I've set up an outbound .call system for customer callbacks and the like. Calls are going out over analog lines and I'm trying to use the WaitForSilence routine to make sure the phone has stopped ringing before starting message playback. The problem is that if I set the first argument of WaitForSilence to anything other than 1, WaitForSilence never exits. Some general info on my setup:

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

Dear All, Just hope your guys out there can help me through..since I've been playing for serval hours....and still not able to resolve it... The workflow as I've created an .call file for Asterisk to call out and it's working fine with outdial, passing variable to asterisk..But the problem is when the calls reached Context and execute AGI script, the script didn't get any

Hi everybody, This is the console output of the asterisk server. debian-te410*CLI> sip set debug peer 2002 SIP Debugging Enabled for IP: 172.26.48.113:5061 I have a sofphone with user 2002 registered on the server on the ip 113. I am trying to place a call to the sofphone on this ip. I have written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have

Exactly. Message: 8 Date: Wed, 08 Feb 2006 13:41:29 -0600 From: "Kevin P. Fleming" <kpfleming@digium.com> Subject: Re: [Asterisk-Users] Re: Need to retrieve Call-ID from dialed SIP channel (w/o CDRs) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43EA4969.60603@digium.com> Content-Type: text/plain;

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

Hi, To combine two sources of CDR's I want Asterisk to save the SIP callid for all calls. I know there's a variable that contains the SIP CallID value, but is this the callid value of the incoming INVITE message or the outgoing message? Are they the same? (I've not yet checked a trace, I'm sorry for that). I've tried to read chan_sip, but couldn't find something in the

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi all, with an (attended) transfer i see the following happening: 1) A calls B1 2) B2 calls C 3) B2 transfers call to A 4) A talks to C At step 3, the channel A is connected to channel C and B1 and B2 are hung up. In the h extension for channel B2, the channel is renamed to B2<ZOMBIE> and i see that the channel variables of A have been merged into B2<ZOMBIE>. If there were

Hi all, I have the below peace of my AGI script...the problem here is that I cannot fetch the extension value to inside the script and assign it to another variable...I highlighted it in red #!/usr/bin/perl #use DBD::mysql; use DBI; use DBD::mysql; use Asterisk::AGI; ############################ #To read asterisk variable values. $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse();

I haven't managed to work this one out yet, so any assistance appreciated ... We want to be able to set the outgoing caller-id on, BRI according to the extension but haven't worked how with asterisk ? we have several hundred inbound numbers on these BRI so we are able o use any one these to sett on outdial. One other point I have been told should work, bu have no way of trying... In

Hi All, I'm trying to configure a Digium T100P to talk to a legacy voicemail system. I have the signaling specs verbatim from the original manufacturer documentation as follows: [T1 Signaling] Service Type: T1,D4 format, AMI(Super Fram) Signaling: Four wire, terminated, E&M (Robbed bit) Start Protocol: Wink start; 250msec duration Dial Tone: Enabled Digits: DTMF, 4-digits DTMF: 50msec

This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that