asterisk

asterisk

can't connect to my Asterisk server with jssip library, Wrong Password error

Can Kamailio pass through http request to asterisk?

Asterisk increase timeout between dtmf tones

a2billing step rate not taking the right time?

How long Asterisk wait for an AGI call?

Asterisk using Customer CDRs with cdr_adaptive_odbc is not working

What can I use to integrate Push-to-Talk functionality in the chat module of my application

a2billing sip invite wrong format?

Sniffing trunked IAX2 traffic

How to configure Elastix inboud route using callerId?

Asterisk AMI set DND ON and OFF programatically

Identify DTMF is pressed or not on Asterisk IVR

Asterisk: Change the IP of PBX, and the phones do not register anymore

FreePbx B2BUA - Skip confirm call

Asterisk 13, AMI, Queue, NewChannel Local

Asterisk unable to find master config file

cel_pgsql Inserting duplicate columns

Retrunning data from VXML to Asterisk dialplan

Asterisk IVRS setting

Asterisk how to detect out of service and mailbox

Run Asterisk AGI when the call is initialized

Trigger a script in a FreePBX Distro when arbitrary call is picked up

Asterisk IP-PBX: API to set up and tear down a call between two extensions

Why i can't able to call python or php code from vxml in asterisk

JSSIP not connecting to FreePBX

Asterisk run script on Phone Answer and Outgoing Call

Asterisk won't does as extension_custom.conf says

Play 2 audio files on Music on Hold interchangeable

How can I run a dialplan in Asterisk through AMI?

Duplicate channel variables after transfer in Asterisk

PJPROJECT building error?

How to insert the call-id through Asterisk Call-file?

Static build of Asterisk 11 on Debian 8 failing

Accept DTMF in ongoing Conference Using Confbridge

How to implement BFCP for screensharing using SIP

Can I set the asterisk_version string

Asterisk get callerid for callback

Playback an audio file to a specific channel in asterisk dialplan

How to edit NAT settings for chan_pjsip

How do I prevent or catch "Unable to run: socket is null" from AsterNet?

Asterisk context configuration

can i disable channel event logging records in database for specific context ?

Asterisk error: Failed to authenticate on INVITE to - can't call

Extremely poor recording quality - Asterisk 11

Asterisk exten Playback sound for different dialplan

Snom 7xx devices bulk setup through FreePBX with custom template

In Asterisk, should I worry about reloading configuration files during calls?

Asterisk (Elastix 4.0.0) Misc Destination Call Hangup Issue

DTMF digits ignoring by option?

Asterisk Playback with no answer on dahdi channel?

Continue Asterisk Dial Plan after Wild Card Match

asterisk error:chan_sip.c:4274 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

Asterisk how to dynamic allocate sip account to ip phone

Asterisk TLS failing to work

Use asterisk as passive recording server

Is group calling over SIP possible with asterisk?

Asterisk own function - execution time [closed]

SIP Registration on asterisk

Call not audible via flowroute with asterisk

Asterisk call termination

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