Asterisk Dialplan: Asterisk dial application (Latest Versions)

The Dial() application is the core of real call handling in Asterisk. It is the command that actually rings another device, connects two people, bridges audio, and creates a live conversation.

Without Dial(), Asterisk can answer calls, play audio, or collect digits — but it cannot connect callers to agents, customers, or external numbers.

This guide explains how Dial() works in modern Asterisk using PJSIP, including syntax, real production examples, and the most common troubleshooting scenarios.

Basic Syntax of Dial()

The general structure of Dial() is:

Dial(technology/target,timeout,options)

• technology → channel driver (PJSIP in modern Asterisk)
• target → extension, trunk, or endpoint to call
• timeout → how long to ring (seconds)
• options → behavior controls (recording, transfers, music, etc.)

Example:

Dial(PJSIP/1001,20)

This rings extension 1001 for 20 seconds.

Modern PJSIP vs Legacy SIP Syntax

Old tutorials often show:

Dial(SIP/1001)

That uses the deprecated chan_sip driver.

Modern Asterisk must use:

Dial(PJSIP/1001)

Always confirm your system is PJSIP-based before copying examples.

Dialing an Internal Extension

[from-internal]
exten => 1001,1,Dial(PJSIP/1001,20)
 same => n,Hangup()

Flow:

• Asterisk rings extension 1001
• If answered → call is bridged
• If no answer → continues to next priority

Dialing Multiple Extensions (Ring Group)

You can ring multiple devices at the same time:

Dial(PJSIP/1001&PJSIP/1002&PJSIP/1003,25)

Behavior:

• All phones ring simultaneously
• First person to answer gets the call
• Others stop ringing automatically

Dialing an External Number via SIP Trunk

External calls typically use:

Dial(PJSIP/9876543210@mytrunk,60)

• 9876543210 → number being dialed
• mytrunk → trunk endpoint defined in PJSIP

This is the foundation of outbound calling and autodialers.

What Happens Internally When Dial() Runs

• Asterisk creates a new outbound channel
• Attempts signaling to destination
• If answered → bridges RTP audio between both channels
• If failed → returns control to dialplan

Understanding this flow is key for debugging real call failures.

Important Dial() Options Used in Production

1) Play Music While Ringing

Dial(PJSIP/1001,20,m)

Caller hears music instead of ring tone.

2) Allow Call Transfers

Dial(PJSIP/1001,20,tT)

• t → called party can transfer
• T → caller can transfer

3) Call Recording with MixMonitor

exten => 2000,1,MixMonitor(/var/spool/asterisk/recordings/${UNIQUEID}.wav)
 same => n,Dial(PJSIP/1001,20)

Handling No Answer or Busy Cases

After Dial() ends, Asterisk sets a status variable:

${DIALSTATUS}

Common values:

• ANSWER
• BUSY
• NOANSWER
• CHANUNAVAIL
• CONGESTION

Example handling:

exten => 3000,1,Dial(PJSIP/1001,20)
 same => n,GotoIf($["${DIALSTATUS}"="NOANSWER"]?voicemail)
 same => n,Hangup()

exten => 3000,n(voicemail),Voicemail(1001@default)
 same => n,Hangup()

Most Common Dial() Problems and Real Fixes

No Matching Endpoint

• Wrong endpoint name in Dial()
• PJSIP config not reloaded

Call Rings but No Audio

• NAT or RTP port issue
• Firewall blocking UDP range

Immediate Hangup

• Codec mismatch
• Trunk authentication failure

Always debug with:

pjsip set logger on
rtp set debug on

Real-World Production Dial Flow Example

[inbound-main]
exten => s,1,Answer()
 same => n,Playback(welcome)
 same => n,Dial(PJSIP/1001&PJSIP/1002,30,tT)
 same => n,Voicemail(1001@default)
 same => n,Hangup()

This single block powers thousands of real business phone systems.

Key Takeaway

Dial() is the heart of Asterisk call routing. Mastering Dial() means you can build:

• Inbound routing
• Outbound dialing
• Ring groups and queues
• Autodialers and AI calling

In modern deployments, always use: PJSIP-based Dial syntax, proper NAT handling, and DIALSTATUS logic for reliable call flows.