[asterisk-dev] Various things

[Johansson Olle E]

4 sep 2006 kl. 10.41 skrev 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.
>
>

Check the SIPCHANINFO and CHANNEL dialplan funcs

> 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...
>
check the SIPDTMF variable

> 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).
>
It is not db only, it's a storage mechanism that also supports LDAP  
and could support what you describe.

> 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...
>
Realtime is not only authentication... There's a module called  
res_auth that you should look into. It's available
in the bug tracker...

/Olle