asterisk

DTMF digits ignoring by option?

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

Using Asterisk AGI to make outbound calls using Originate and controlling both sides of call

Recording ChanSpy in Asterisk

SIP Registration on asterisk

Asterisk Getting stopped when handling more than 200 calls

Call not audible via flowroute with asterisk

How to identify Asterisk SIP Messaging delievered and seen status?

Protocol to Hang up from browser on any softphone

Call relaying via GOIP with callee number

Freeswitch does not respond to repeat request 407

install asterisk web interface

How to put call on hold via cli or ami or agi

Asterisk - playing music whilst originating a call

asterisk Call ID in sipml5

Asterisk: Can call but can't hear voice

Call Transfer, do not disconnect unless connected

SIP invite is not received until 180 seconds

Asterisk - Transfer call record not saving in database and recording filename not overriding

Local SIP to Local SIP Speed on “hosted pbx”

Asterisk Music On Hold with Time Conditions

Asterisk TLS failing to work

How ISDN card connects with PSTN

Asterisk, two IP address, same segment, same phisical interface

Is group calling over SIP possible with asterisk?

How can I direct calls to my Verizon Wireless Voicemail using SIP?

Asterisk silence until agent answers

Asterisk Sipml5

How can I remove noise in asterisk PBX?

Asterisk Call File Call failed to go through

Asterisk own function - execution time [closed]

Asterisk - how to execute script on call received

Call forwarding GOIP16 with asterisk

Asterisk call termination

Voice message is getting blurred when asterisk receive more than 180 calls

Asterisk 13 - Call Transfer Agent to Agent Macro

Asterisk returns 481 on hangup

How to set up a VoIP server/service (Asterisk) on AWS with a link to phone lines?

Incoming call does not work

Enable chan_mISDN in asterisk 1.8

Automatic terminating a call when call made

limit number of watchers in asterisk BLF/subscriptions

How to allow inbound calls in pjsip and Asterisk 13?

auto answer when catching the phone with asterisk

SIPML: Error while remote call is coming

Asterisk 13.10.0 does not detect user input. DTMF not working

Simultaneous application execution

Asterisk Event to listen to for Transferred call

can not hear playback sounds ( asterisk )

get the caller number after call is answered

Asterisk automatic incoming call answering

Error Asterisk 12.8 bad magic number

Can not configure adhearsion with asterisk

Asterisk WAN Issues

Unable to establish two way Asterisk call

Asterisk GotoIf comparison variable exists in Array

Call Limit (call-limit) in Asterisk 13 with PJSIP

Asterisk call recording not showing any logs and file

Asterisk 13.7 GROUP_COUNT always returns 1


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