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Questions tagged [freepbx]

FreePBX is an open-source web-based GUI for Asterisk (PBX), a voice over IP server. On-topic questions include those about writing FreePBX modules, or extending built-in dialplan. Stack Overflow is not an appropriate place for server management or general support of either FreePBX or Asterisk.

FreePBX allows easy management of the Asterisk PBX and automates the creation of complex applications such as IVRs and conferencing systems, as well as management of user extensions and features. It is comprised of various modules which can be independently installed or uninstalled, and an API is provided for third parties to create their own modules.

FreePBX also provides a Linux distribution based on Red Hat Enterprise Linux 7.0, which provides a pre-installed version of FreePBX that is able to run commercially licensed modules.

187 questions
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FreePBX Twilio Outbound Ringtone

I have a Twilio SIP trunk connected to FreePbx, all users are using the webrtc module of FreePBX to make calls. They can make and receive calls fine with two way audio, however with outbound calls the caller does not hear ringtone (ringing) as the B…
Freddy Wetson
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AsteriskNOW vs FreePBX. What is the easiest to customize?

I develop asterisk and GUI. Asterisk GUI were exist several type. FreePBX, AsteriskNOW, Elastix, Trixbox... Finally, I have selected two type. FreePBX and AsteriskNOW. FreePBX is based on php, AsteriskNOW is based on java. Almost people used…
whdals0
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Asterisk Dialplan: How to detect when a call has been successfully answered?

I'm having a really hard time figuring out if there is a trigger or a way to continue from the Dial action that allows you to detect if the call is answered. It seems like Dial doesn't respond until hangup, busy, or congested. What action or event…
Jb1128
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Asterisk IVR After Hangup

I want to redirect caller to an IVR after dialed number's hangup. I made research and found something called deadAGI but I couldn't make it work. You can find my extensions_custom.conf file below. [from-internal-custom] exten =>…
Deniz B.
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Send DTMF Digit with dialplan

I have the following dialplan: exten => 224,1,NoOp(IN CALL : ${CALLERID(num)} => ${EXTEN}) exten => 224,n,Dial(${PJSIP_DIAL_CONTACTS(97,97)}) ;exten => 224,n,Wait(3) exten => 224,n,SendDTMF(*11234*,200) exten => 224,n,Wait(3) …
wurzeldd
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How to redirect already answered incoming call using pjsip?

I use pjsip.dll for creating softphone app. Answering and dialing works fine. Now, I need to redirect already answered incoming call to another sip-user (for example, from number 101 to 104). How to do that? I cannot find function in pjsip sdk…
РСИТ _
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Originate call to sip trunk via asterisk manager api java

So I am a total newbie in asterisk and managing call lines in general but I managed to install Asterisk Now 13 distro, I have connected 2 sip phones with pjsip and configured a sip trunk which works when I dial an external number with the…
user4313427
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PJSIP - Freepbx - Trunk Registration Rejected

I'm trying to register a SIP account to my provider. Actually on my FreePBX I have other 4 accounts on different servers registered without problems. I cannot figure out because this specific account don't want to collaborate. This is the log that…
Diego Aguiari
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Channel not in Stasis application in asterisk ARI

I am working with asterisk ARI.My asterisk version is 12.5.0.I have created channels via asterisk ARI.But when i am trying to call it is not display in GUI.I am using zoiper softphone.It showing like below : 409 - Channel not in a Stasis…
Ruchi Patel
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The transmission interval for packets sent to Asterisk with SILK codec is increasing

I'm using a SILK codec with Asterisk. (Asterisk is on Version 11.19) The SILK codec seems to be working correctly, but as time progresses the interval between packet transmissions increases. Because if this, telephone calls are interrupted. A…
kA tA
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channel originate, how to do call from a local channel? (call intercom and send dtmf)

My goal is to : run a background task activated by dynamic feature while in active call, that will execute dial to another EXT and send DTMF. It means, when a user is active call with someone, when the user press 5555, the door will be opened. In…
Yair Glikman
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Hacker makes international calls through my FreePBX IVR

I am running: FreePBX 12.0.76.2 Asterisk 11.18.0 FreePBX 64bit distro 6.12.65 I have many POTS lines for incoming and outgoing calls, and a Twilio SIP trunk for outbound International calls. I just had repeated calls from three different caller…
ITIA
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Calling a PHP script using FreePBX and Asterisk

So I have a VOIP system set up through a FreePBX server. I want to have it so that when a new call is picked up by FreePBX, asterisks will send the caller ID and the call ID to a php script, which will then use that information to gather ticket…
CaptainQuint
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asterisk cli command "channel originate" with call duration or length

I am using asterisk (Freepbx). I am using following command which is working fine. I can originate call from asterisk cli without any issue. channel originate SIP/tunk-name/1416XXXXXXX extension 701@from-internal call ring my phone number…
Fida Hussan
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Freepbx: Add Extension Programmatically

I was trying to add extension programatically. There I was facing some issues. My code was
LuckyCoder
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