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We are using Asterisk 1.4 with Asterisk realtime enabled using a2billing database.
On DID calls the INVITE to the registered client is sent with incomplete codecs when the codec definition in SIP friends is done as
allow=ulaw, alaw, gsm
Internally Asterisk sees codecs with white space in front, as:
"ulaw"
" alaw"
" gsam"
as it does not know a " alaw" codec it ignores those and send INVITE only with ulaw.
The codec definition must always be without whitespace.
IMHO peer-friend section in A2billing.conf should point that out and ideally A2billing should strip whitespace from the allow string.
Cheers
Geejay
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