asterisk

asterisk

iPV 6 changes frequently for sip signalling [closed]

create new a way to php8 agi on issabel [closed]

X-Lite call failed forbidden error while setting up Asterisk telephony [closed]

Asterisk - create a new dialplan file [closed]

How to set dial options in Asterisk callfile

Asterisk, Goto+Hungup

Asterisk ARI Caller id is always Anonymous

variables in extensions_custom.conf

Asterisk extension, conditional execution?

Asterisk Freepbx - hide external number on user's display after forwarding

Asterisk Originate a muted call

How to redirect call from one Asterisk server to another to specific extension? IAX2

How do you write an Asterisk Dialpan to suspend 180-Ringing until after 183-progress in chan_sip?

my execif variable ignored in Asterisk dialplan

Asterisk GOTOIf Command not hitting branch correctly

After recording with mixmonitor, run the AGI script and POST the wav to another server via CURL

Asterisk 11 match destination extension in macro

Asterisk EAGI audio while running AMD or other asterisk app via "EXEC"

Asterisk Hangup caller after answered's hangup

Asterisk use say with new mode

IVR Asterisk Dial Plan goes indefinite for invalid key press

Asterisk GoSub() function not working for me

Asterisk AMI events sometimes missing

What is the Asterisk Dial() Option to Call Subroutine on "Ringing" Status Received from Called Party?

Asterisk - setup hangup after seconds dialplan otpion S(sec)

Play Some Greeting message at first when I dial any extension in Asterisk?

How to make a change in the VoiceMail Asterisk 11

Continue executer next extension of contexte after dial call

Deleting FreePBX Speed Dials

Asterisk sip trunk peer NIC

how to save events ami asterisk in DB

How to allow a skip or goto at any time during a context in Asterisk

Playing Multiple Sound Files

dial a call and add to conference room

Sip call with esp8266 and asterisk

How to make incoming call recors on asterisk?

Asterisk block incoming call from specific country only

Stasis app not active - Asterisk error when dialing extension

Gotoif how to set it to allow a 3 or 4 digit response to meet the condition

Pulling only Registration Contact information on Asterisk

ERROR[12799] netsock2.c: getaddrinfo("ims.ntc.net.np", "(null)", ...): Name or service not known, How can i resolve this issue?

Play MOH instead of ringtone in asterisk

installing asterisk on ubuntu 18.04

Asterisk broadcast mechanism

How to add SipAddHeader for all calls?

Vicidial Group queue to call the next number if the first one is unavailable, busy or rejects the call

Finding uniqueid of call channel

Finding corresponding trunk channel

ERRORS while building asterisk using meta-telephony layer

Asterisk AGI script falls when caller hangup

PJSIP: Why am I able to make calls but logged as unavailable in the Asterisk 16 console?

Asterisk integration with custom ERP

How to define asterisk custom queue ring strategy?

Hang up the call after x seconds after the called party has answered the call using ARI in Asterisk

Asterisk keeps confusing public and private IPs so calls have no audio when softphones are behind NAT

Simplest Asterisk outbound proxy

How can I create hints for extensions created with a 'pattern matching' defintion

Asterisk Sip Reachable timeout

Where Is Asterisk FreePBX Video Call Recording File

Control of SIP phone volume on attended transfer in Asterisk

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