How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk
The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes)
and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk.
So if Phone 1 calls Phone 2 the codec to be g711, but when Phone 1 calls Phone 3 to use g729 ?



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