asterisk users - Jun 2006 - reinvite, DISA, and switching codec's.

My setup is this:

Analogue phone attached to a Linksys PAP2
|
Asterisk
|
VoIP provider

I have put the PAP2 in 'batphone' mode where when you pick it up it
immediately dials the 's' extension in the pap2_incoming context in
Asterisk, where asterisk answers the call and does a DISA(no-password,
internal). I do this because it means I can centralise all of my
dialplan logic in Asterisk regardless of the ATA in use.

Am I right in saying that because Asterisk has Answer()'d the call and
done DISA(...), I can't do a re-invite to bridge the call between the
PAP2 and the VoIP provider? And even if I could, I couldn't set it up to
use G.711a between the PAP2 and Asterisk, and the switch to G.729 when
the call bridges to the VoIP provider? This would be useful in that I
can use G.711a locally where I have the bandwidth, and I wouldn't need
to get licenses to use G.729a (because asterisk wouldn't need to touch
it).

Assuming I'm correct, is this a limitation of the SIP protocol, or a
limit of Asterisk's implementation of it?

Thanks

James

James,
> Am I right in saying that because Asterisk has Answer()'d the call and
> done DISA(...), I can't do a re-invite to bridge the call between the
> PAP2 and the VoIP provider?
Yes, you can reinvite after Dial()'ing your provider, but you probably
won't be able to switch codecs once the call is connected. I may be
wrong so just try it :). The ATA must be able to talk directly to your
provider in such a case (i.e. not NAT).

--Luki