[Asterisk-Users] SIP Codec problem?

[Jim Paraschou]

Hi all,

 I have an * installation with one AVM Fritz card, one
TDM 40b and one sip phone. The problem i have is that 
when i place calls from the sip phone everything works
fine. When i receive calls to the sip phone i cannot
hear and the calling party cannot hear too.
Is it a codec problem ?
Here is my sip.conf.

[general]
port = 5060                     ; Port to bind to
bindaddr = 192.168.0.150                ; Address to
bind to
context = internal
disallow=all
allow=ulaw
tos=lowdelay
maxexpirey=3600         ; Max length of incoming      
 registration we allow
defaultexpirey=120              ; Default length of
incoming/outoing registration
notifymimetype=text/plain       ; Allow overriding of
mime type in NOTIFY
;
[15]
type=friend
username=15
;secret=blah
host=dynamic
defaultip=192.168.0.203
dtmfmode=inband                ; Choices are inband,
rfc2833, or info



 

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