I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything.
Can we please kill "reinvite" - it does not exist in the SIP channel as an option for anything. Period.
There is an option called "canreinvite" that you can set to yes or no. Setting "reinvite" to anything will not change anything at all.
However, setting "canreinvite" to something will change ASterisk's behaviour during a SIP call. It may also break your conversation if your SIP device does not support the SIP re-invite mechanism.
Please read: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+canreinvite for more information.
/Olle
PS. I know that the "reinvite" option is mentioned in many archived e-mails, which does not help at all. Please do not add any more messages with this option, as it will only confuse users.
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