[asterisk-dev] Codec matching

[asterisk_dev]

Hi;

There is something I thing it would be very useful and I see no way for 
doing it with asterisk; To negotiate codecs, usually you want to some 
telephones use g711 when calling to some telephones (for example in the same 
LAN), and g729 when calling others (for example, when going thru WAN). Say 
you have 2 offices, with numbers 111, 112 and 113 for the first one and 
211,212 and 213 for the second. You'll want 1XX numbers use g729 when 
calling 2XX numbers, and use g711 when calling 1XX numbers.

In Cisco CallManager that's what regions are for. Do we have something 
similar in asterisk? If not, it would be a very welcome addition.

Best regards,
  Francisco Sedano.


----- Original Message ----- 
From: "Olle E Johansson" <oej at edvina.net>
To: "Asterisk Mailing List Developers" <asterisk-dev at lists.digium.com>
Cc: "Olle E Johansson" <oej at edvina.net>
Sent: Saturday, March 04, 2006 10:57 AM
Subject: [asterisk-dev] *** Yet another boring weekend - updates to the 
testbranch! *** TEST NOW!


> Join the cool crowd that tests the test branch during evenings and 
> weekends. The dudes and dudettes that
> proudly contributes by reporting everything from simple spelling  errors 
> to crashes and strange noices from
> their Asterisk boxes. The people who knows what is going on in the 
> Asterisk development circles - the
> Asterisk Test Team!
>
> I've updated the test branch to the latest version of my SIP  peermatch 
> code. This is quite a large code
> change, but not as large a functional change. However, it changes  some 
> basic functionality:
>
> * The sip_user structure is gone
> * Incoming calls are matched first by user from: name, then peer  From: 
> name, then IP.
> * Friends are now *one* in-memory object.
>
> In most cases, this means you can change type=friend to type=peer for 
> local phones on the
> same LAN. This will also improve SIP subscriptions (blinking lights)  and 
> call limits, since for
> both friends and peers, we now have *one* object in memory that  handles 
> the limit for both incoming
> and outgoing calls.
>
> During the week, I've also added a few other patches by other 
> contributors.
>
> Read the README.test-this-branch here:
> http://svn.digium.com/view/asterisk/team/oej/test-this-branch/ 
> README.test-this-branch?view=markup
>
> ** PLEASE help the community, please test this branch.
>
> Check it out like this
>
> svn checkout http://svn.digium.com/svn/asterisk/team/oej/test-this- 
> branch test-trunk
>
> Then cd into test-trunk and run "make" then "make install"
>
> Report any bugs in the proper open bug in the bug tracker. If you
> like new functions, add a comment that this works for you. Provide
> feedback, make our work easier.
>
> Run "svn update" from time to time to get the latest version. Any
> changes from trunk will be merged into this code. Read the
> README.test-this-branch file to get more information.
>
> Thank you for your help!
>
> /Olle
>
> PS. Obviously, this is test code, not recommended to be closer than 2
> miles from your production servers.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>