Posted: Fri Mar 28, 2003 1:46 pm Post subject: [Asterisk-Dev] mgcp<>sip dirty fix
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.
Posted: Fri Mar 28, 2003 2:23 pm Post subject: [Asterisk-Dev] mgcp<>sip dirty fix
Quote:
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.
Posted: Sat Mar 29, 2003 1:04 am Post subject: [Asterisk-Dev] mgcp<>sip dirty fix
Quote:
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).
Posted: Sun Mar 30, 2003 5:20 pm Post subject: [Asterisk-Dev] mgcp<>sip dirty fix
Quote:
> 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.
Quote:
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).
But in this case, INVITE wasn't sent by asterisk (i.e. Asterisk is the
callee, not the caller). Is it still permissible to send INVITE?
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