Joshua Colp wrote:
We have a customer who would like to do RTP directly between SIP
devices. The devices are not registered directly to Asterisk, but to SER
on another machine.
It seems in this case "canreinvite = yes" is never used. Does anyone
know of a way of persuading Asterisk to issue re-invites in this case?
What do you mean by not used? Even if going through SER it should still be used.
Joshua,
That's what I would have expected, but Asterisk is not issuing re-invites.
One thing I should have mentioned is that Asterisk, SER, and the phones
are all on an RFC1918 network (but there is no NAT between them). Maybe
Asterisk is seeing that:
1. The phones are offering an SDP address that is in RFC1918.
2. The SDP and SIP headers are not the address the SIP is coming from,
as SER is relaying them.
and therefore incorrectly concluding that there's NAT between it and the
phones?
Alistair Cunningham,
Integrics Ltd,
+44 20 799 39 799
http://integrics.com/
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