[Asterisk-Users] SIP/ Grandstream Issues

[Uriel Carrasquilla]

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 at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Lists
Sent: Saturday, September 27, 2003 7:01 PM
To: asterisk-users at 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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