Asterisk sip progressinband

sip.conf: progressinband


When "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio


Send 180 Ringing if 183 has not yet been sent establishing audio path. If audio path is established already (with 183) then send in-band ringing (this is the way asterisk historically behaved because of buggy phones like polycom)


Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behaviour of Asterisk.


Note that inband progress is not usually desired, because it requires extra Asterisk resources to run a generator to generate the inband ringing. If the endpoint is generating its own, then there is no need to tell Asterisk to do it.

Once '183 Session Progress' is sent, it is not useful to send '180 Ringing' any longer... the 183 message informs the endpoint that future progress indications for this session will be provided inband via the audio stream. Most (if not all) SIP endpoints would ignore any 180 received after a 183 anyway.

WARNING: To get inband audio on asterisk 1.6.2.x it is sometimes necessary to set prematuremedia=no. If instead prematuremedia=yes(default) is set, then inband will not work

Related ISDN example

Q: I use 2 ISDN channels with a with a fritz! card and the junghanns capi drivers. The problem appears with SIP to ISDN calls.
The SIP 180 ringing message doesn't appear because the ISDN PBX sends the "ALERT" message in-band (channel B), and not in the D channel. So Asterisk doesn't know when the ISDN channel is ringing. With my configuration Asterisk can not understand the in-band signalling for the capi channels, is it possible to use "in-band" signallisation for capi channels?

A: I guess that's "Early Media Connect", i.e., if the phone supports that (not all do), the channels get bridged just after dial completed, (SIP 183), and what you hear is the remote ring tones (from your telco), not locally generated (as if it received SIP 180 Ringing).

See also

Go back to Asterisk config sip.conf

Created by: JustRumours, Last modification: Sun 06 of Jan, 2013 (17:45)
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