Try on the Grandstream DTMF via INFO.
Also use uLaw for codec.
If behind the NAT just say NAT=YES and REINVITE=NO.
It works like a champ.
Regards,
Uriel
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Lists
Sent: Saturday, September 27, 2003 7:01 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP/ Grandstream Issues
I just got a grandstream SIP phone
Here is my sip.conf for the phone
[mlh]
type=friend
insecure=yes
username=mlh
secret=mlh
host=dynamic
canreinvite=no
The phone as the default config on it.
If I use the phone to call a Zap interface (a tdm card) the voice sounds
all choppy.
If I use the phone to call a x100p card, it does not dial what I dial (no
DTMF)
I don't know what else to try.....should I change the vocoder (it is on
PCMU at the momemnt)
I am using the phone on a LAN so bandwidth is not an issue.
Any Help would be great,
Michael
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