[Asterisk-Users] Codec problems??? (Was: SIP i.e. Is something broken?)

[Clif Jones]

I was looking at some fixes in the replies to the chan_sip.c problems and
I am wondering if I am seeing the same thing in the earlier file version.  I
just checked to see that my chan_sip.c is version 1.179 when I did my
checkout so I never had the later versions.  The problem that I am seeing
is that DTMF is not going from 1 SIP device to another and sometimes
voice is not going from 1 SIP device to another.  Most of the SIP endpoints
are Cisco 7960's running version 5.x firmware and the SIP Gateway is
an AudioCodes 4-port SIP FXO.  The missing voice problems seem
to be coming into play when we use X-Lite.  I have tried disabling the GSM
codec and going only with G7.11 but have not found the right recipe to get
DTMF from end to end and voice from X-Lite out through the Audiocodes
gateway.  Anybody have any ideas or know what debug levels I need to
turn up to see the codec negotiations?  My suspicion for the DTMF problem
is that the Phone-Event codec is being accepted by Asterisk but it is 
somehow
getting lost between Asterisk and the Gateway.  I would like to verify this.