[Asterisk-Users] codec problems with astcc and not with sip trhough aix

[sjaak nabuurs]

Hello .

I have a 800 tollfree trhough iax to my * server.
If I phone to 800 number to the * machine to a  sip phone everything is 
okay.
exten => 8XXXXXXXX,1,Dial(SIP/12345678,20)
    -- Accepting AUTHENTICATED call from xx.xx.xx.xx, requested format = 
4, actual format = 4

If I change my extention to the astcc script i get lott's of these errors
exten => 8XXXXXXXX,1,DeadAGI(astcc.agi)
Oct 21 00:58:22 NOTICE[1110648512]: channel.c:1314 ast_read: Dropping 
incompatible voice frame on IAX2/domain at xx.xx.xx.xx.186:4569/1 of format 
G729A since our native format has changed to ULAW

I see these two difference are codec problems but why.
and what to do with this problems

When I phone with a second SIP phone to the 8XXXXXX number everything is 
okay to the sipphone and to the astcc script
My sip phones accept "all" codecs

My aix.conf looks like this
[tollfreeprovider]
type=user
host=xx.xx.xx.xx
secret=mysecret
notransfer=yes
disallow=all
;allow=g729
;allow=gsm
allow=ulaw
trunk=yes
context=tollfree

I don't know, could this be a problem for my tollfree number provider or 
is it me.

Cheers