asterisk

asterisk

How to handle channels into an ARI queue app? [closed]

How to execute code before making a call on Asterisk

Asterisk conference call works with Polycom phone, but how to make it work with microsip soft-phone

Asterisk: How to implement outbound rate limiting (calls per second)

Asterisk CHAN_SIP update custom contact host URI

Monitor asterisk calls with MRTG

X-Lite Asterisk Failed to establish call - Forbidden number

About Asterisk VOIP

Asterisk issabel auto park

Asterisk Refer to new IP address

Error in Insert Value "answer" in CDR Database Asterisk

Asterisk BridgeWait How to bridge into a general bridge

When should i use $() in asterisk syntax?

How to recall in originate command with dialplan?

Asterisk do not terminate channel

How to buffer DTMF in Asterisk?

Sip status (ringing, busy, in use, etc.)

How to log unanswered call? Ubuntu 18 + Asterisk 16 + FreePBX 15

How to detetct inboud abandoned calls in a queue

Incoming calls show as answered, but the caller is dropped

1 way audio only when registering to OpenSIPs in front of Asterisk

Asterisk SIP - Dial in parallel

Asterisk do not sees queue members

Asterisk 11 Who Hanged Up?

Get how long device on hold state in call

Freeswitch Guide how to get CDR UUID from client side

Why Asterisk answered "Bad Event" to PUBLISH Event: capability-change?

Is there any function in confbridge application to mute audio between users?

How to add event listener to call hangup PHP AGI?

Asterisk AsterNET How to move from parking to queue?

Asterisk Make Menu Select mysql_odbc option is disabled

By default, asterisk will generate a CDR for every finished call in database cdr table

How to call an another phone after hangup the first?

Missing FilterList in Asterisk AMI (manager)

Asterisk returns 501 - Not Implemented to INFO packets

Asterisk - Make a call inside a conference with ringtone

Voximal: Unable to connect to UniMRCP compiled with custom ASR plugin

Asterisk (version 16) Redirect action does not complete 1st leg of call

Asterisk SendURL() using grandstream UCM6204

How do i pass multiple variables to writing ODBC function?

Can we port Asterisk in rtxRTOS

Increasing timeout on channel originate

Asterisk, Ignore DTMF special characters in GET DATA

"Please Select the Default Locales of the PBX" page comes back fairly often in FreePBX

${STRFTIME(${EPOCH},,%d/%m/%y %k:%M)}

How to limit the number of call record in CDR file

How to get caller id variable?

Queue calls strategy

Push asterisk data to X-lite Softphones

Call STT based on customer response

Asterisk, Dial plan, how can I hang after answer?

Asterisk one way audio

Why does Asterisk ignore a dtmf code from a specific handset

Using Asterisk ARI to determine if agent in queue pause or not

In asterisk i want to allow only 10 digit caller ids to dial through the system

Googl ASR integration using UniMRCP for two way communication

Differences and best protocol for voip asterisk pjsip

WaitExten() problem to identify typed numbers

Multiple asterisk servers SIP or PJSIP realtime

input of an arbitrary number by the user without long waiting for the extension

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