Early Media with Progress()

Progress() sends a SIPSession Initiation Protocol, the standard signaling protocol used to set up, manage, and tear down VoIP calls between Asterisk and phones or carriers. 183 Session Progress response with SDP, establishing a media path to the caller before the call is answered. This lets you play audio (custom ringback, announcements, or music) while the call is still in the "ringing" state.

Dialplan

[from-trunk]
exten => 1001,1,Progress()
 same => n,Verbose(1,Early media: playing greeting before answer)
 same => n,Playback(custom/please-hold)
 same => n,Answer()
 same => n,Dial(PJSIP/1001,30)
 same => n,Hangup()

Progress vs Answer

[examples]
; Early media -- caller hears audio but call is not "answered"
; (no billing on metered trunks, CID shows "ringing" on caller's phone)
exten => 2001,1,Progress()
 same => n,Playback(custom/welcome)
 same => n,Answer()
 same => n,Dial(PJSIP/2001,30)
 same => n,Hangup()

; Immediate answer -- billing starts, phone shows "connected"
exten => 2002,1,Answer()
 same => n,Playback(custom/welcome)
 same => n,Dial(PJSIP/2002,30)
 same => n,Hangup()

How it works

1. 183 Session Progress: Progress() causes Asterisk to send a SIP 183 response with an SDP body. This establishes a media (RTPReal-time Transport Protocol. Carries the actual audio (media) of a VoIP call after SIP signaling has set it up.) path so the caller can hear audio, but the call state remains "ringing". the caller's phone still shows "calling..." or "ringing."
2. Billing implications: On metered SIP trunks, billing typically starts when a 200 OK (Answer) is received. Using Progress() for pre-answer announcements avoids starting the billing clock.
3. Playback before Answer: After Progress(), Playback() sends audio over the early media channelA single call leg passing through Asterisk. Channels represent connections to endpoints and are what dialplan applications act on.. This is how carriers play "your call is being connected" messages or custom ringback tones.
4. Answer() still required: Early media is temporary. You must still call Answer() (which sends a 200 OK) before bridging the call with Dial(), Queue(), or other applications that expect a fully established call.
5. SIP signaling flow: The sequence is: INVITE -> 100 Trying -> 183 Session Progress (with SDP) -> [audio plays] -> 200 OK -> ACK -> [call is bridged].

Tips

• Not all SIP providers and phones support early media correctly. Test with your provider before relying on it for production IVR greetings.
• Use Progress() + Playback() for "please hold while we connect your call" messages before Dial().
• If the caller's phone doesn't support early media, they'll hear nothing or their carrier's default ringback until Answer().
• For inbound calls from the PSTN, early media is well supported. For calls between SIP endpoints on the same PBXPrivate Branch Exchange, a private telephone switch. Asterisk is a software PBX that routes calls between internal extensions and the outside world., it depends on the phone firmware.
• Ringing() sends a 180 Ringing without SDP: the caller hears their phone's built-in ring tone. Progress() sends 183 with SDP. the caller hears audio you provide.