[Asterisk-Users] Grandstream Early Dial

[David J Carter]

Hi,

I have my GS set to in-audio for DTMF and as bellow for my sip.conf: -

[7001] ; SIP Phone
type=friend
insecure=yes
host=dynamic
reinvite=no
canreinvite=no
nat=1
mailbox=7001
dtmfmode=inband
callgroup=1
pickupgroup=1
disallow=all
allow=ulaw
allow=alaw
allow=gsm

I am using 1.0.4.26 and all is working fine.

The only differance I have noticed since moving up to 1.0.4.x is the speaker
volume is lower on speakerphone.

Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Dave Cotton
Sent: 31 December 2003 07:13
To: Asterisk List
Subject: Re: [Asterisk-Users] Grandstream Early Dial


On Wed, 2003-12-31 at 05:53, Tilghman Lesher wrote:

> What happens when you change the configuration of the GS phone to
> send DTMF via SIP INFO?

I've just checked my voicemail with 1.0.4.30 and get the same multiple
digits problem. sip.conf and GS config are both at info, for me this is
a new problem voicemail has always worked perfectly with the GS.

I can't go back to 1.0.3.81 to remove that variable. I updated from CVS
on the 26th.

--
Dave Cotton <dcotton at linuxautrement.com>

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