Hi,
first let me explain my configuration:
astlinux has ISDN PCI card and SIP trunk over ADSL.
Extensions, analog phones, are connected to asterisk over Linksys PAP2T.
My question is about codec. When I call local PSTN numners over ISDN asterisk
should use alaw codec, but if I call someone over SIP trunk asterisk should use
g729 codec. Both Linksys PAP2T and SIP trunk support g729 codec.
Can this be achieved with the usual asterisk configuration or I should
experiment with Asterisk Codec Negotiation Patch:
http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch
Regards
Dominko
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