[asterisk-dev] Codec matching

[mirza sahib]

top posting madness continues ...

You mean something like SetCodec(value), where value is from show codecs, 
so something like

 	exten => 1XX,1,SetCodec(5); would allow alaw and ulaw


possible, folks?

- wasim



On Sat, 4 Mar 2006, Alberto Sagredo wrote:

> 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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