[asterisk-biz] Asterisk SIP deleyed offer

[Dmitriy A Pyryakov]

Hello, I have problems with calls between asterisk and avaya with canreinvite=yes enabled and Direct IP-IP? y (on avaya).

Direct rtp work well but when avaya side client press hold,transfer or conferens, avaya use deleyed offer (INVITE without SDP). Asterisk don't resend this INVITE to client, just send 100 and 200 ok with SDP and get back ACK with new session SDP from avaya side (it use medpro board for music on hold). Asterisk don't send this information to client softphone because invite from avaya without SDP. Avaya send rtp from medpro to asterisk softphone and asterisk softphone still send rtp to avaya side phone directly.
After avaya phone release hold button, avaya send again INVITE w/o SDP, asterisk  send 100 and 200 ok to avaya, and then send INVITE to asterisk's softphone with SDP, but this SDP it gets from previous ACK. Actual Information from avaya backed with ACK again, but asterisk don't tell about it to his softphone.

I hear that cisco system have some special settings to fix it by enabling early offer. Avaya - don't.

Is it possible force asterisk to use delayed offer?

Sorry for my English.


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