[Asterisk-Dev] mgcp<>sip dirty fix

Mark Spencer markster at digium.com
Fri Mar 28 18:04:48 MST 2003

> I don't understand enough of asterisk's SIP implementation to understand
> what should properly happen in that case, but the behaviour you would see
> is after call is picked up, * would issue another INVITE to the other end,
> and it confuses the hell out of any SIP stack.

It is entirely permissible to send another invite with an updated SDP.  It
seems that some devices (e.g. ATA-186) don't handle it properly (or there
is somehow something wrong with our re-invite).


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