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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

Hi folks I'm not sure what I am missing but I cannot find a predefined channel variable to identify the SIP peer/user which has initiated a call and established the channel. The one option is to extract it from the CHANNEL variable, but that is fraught with difficulties. Is there another variable I don't know about or another way to do this? Thanks in advance! Richard -- Richard

Hi folks I have a handset talking to Asterisk, which in turn puts the call through to an ITSP. The handsets likes to send audio in 40ms frames (even though Asterisk requests 20ms frames with a ptime header in the SDP). The ITSP doesn't request any particular frame length (with ptime) or state a maximum length (with maxptime), so when Asterisk receives the 40ms media frames from the handset,

[Resent due to non-descriptive subject line.] 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? Scenario as follows: 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

Hi all If Asterisk is bridging a call between two SIP peers and one peer puts the other on hold by means of a re-INVITE with SDP containing a=sendonly, Asterisk will play locally generated MOH instead of relaying the media streamed by the SIP peer which took the hold action. Any ideas how to change that? (This is understandable if the peer is a handset but can be a problem if it is a PBX with