[Asterisk-Dev] mgcp<>sip dirty fix

[alex at pilosoft.com]

> Alex what's the symptoms this fix cure, I'm having problems with mgcp ->
> sip as well, my mgcp phone will initiate a call behind asterisk travel
> sip to my cisco pstn gateway, I get one ring and it drops basically.
> MGCP -> MGCP works fine, and zap -> mgcp, and mgcp -> zap work fine, sip
> -> zap, and zap -> sip seem to be fine as well.
Yep, that's the symptom. And the patch fixes it. It probably breaks 
something else, though. I'm sure reinvites are necessary at some 
situations (when the encoding changes or something).

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.

-alex