[asterisk-dev] Codec matching

[Alberto Sagredo]

As i know, SPA941 has this option inside.

Use g711 when LAN, and selecting 729 for other stuff.

Anyway codec selection in asterisk has to be improved a little bit more.

I hope this this helps.

asterisk_dev escribió:
> 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.
>> _______________________________________________
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> 
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