[asterisk-dev] Various things

[Jean-Michel Hiver]

Hi List,

There's a few things I would need from Asterisk, so I was wondering if 
there was a way to hack the source to have them. These would be:

SOURCE_FROM: IP address of the party which initiated the call. Could be 
empty for non voip channels.

SOURCE_CODEC: Codec which has been negotiated for the party which 
initiated the call. This would allow to route only to gateways which 
support this codec.

How hard would it be to dynamically change the preferred codec order? 
Say I accepted ulaw alaw g729, peer a elected to use g729. I want to 
change the codec order of peer be to place g729 as a preferred codec, to 
avoid doing transcoding wherever possible...

How does Asterisk negotiates codecs when no codec order is specified 
(i.e. allow=all)?

Regarding realtime, why make it DB specific only? IMHO it would be nicer 
to have a network interface so you could write a server to authenticate 
against anything. It could use some well defined XML format for data 
exchange (which is what XML is actually for).

For example, Perl has a pretty cool module for writing servers called 
Net::Server and a plethora of modules to authenticate against radius, 
databases, PAM, etc...

Same goes for dialplan... a network interface would have been much nicer 
IMHO and would be very useful for LCR engines for instance.

Cheers,
Jean-Michel.