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[Richard Brady]

Hi folks

When connecting two SIP users, is there any way to stop Asterisk from
sending SIP 183 Session Progress messages, either globally or
per-peer?

Call from UA1 to Asterisk (UA2) to UA3
UA3 sends RTP before SIP OK to Asterisk (UA2)
Asterisk (UA2) detects early audio from UA3 and sends 183 Session
Progress with SDP to UA1.

Instead I would like it to just send on the early audio, is this possible?

Thanks in advance,
Richard