[Asterisk-Users] Grandstream 101

[Steven Ringwald]

dkwok wrote:

> Just got GS 101 phone and plugged into the network.
>
> Got ip setup however, the following problems arise:
>
> 1. when dialing an extension, I cannot further send any key tone to 
> Asterisk.
> 2. there is no sound coming from the other end.
>
> I have a sip.conf setup for GS:
> [General]
> disallow=all
> allow=ulaw
> allow=alaw
>
> [gs]
> canreinvite=no
> dtmfmode=info
>
> In the GS101 setting
> rtp port = 5004
> sip port = 5060
> dtmf = sip info
> codec = pcmu
> codec = pcma
>
> Any pointer of a sample of config file would be most appreciate.
>
Here is what my sip.conf file looks like for a grandstream phone:

[sringwald]
disallow=all
host=dynamic
allow=ulaw
type=friend
username=sringwald
secret=<SOME SECRET>
callerid=Steve <777777>
canreinvite=no
reinvite=no
insecure=yes
nat=yes
dtmfmode=inband         ; Choices are inband, rfc2833, or info
mailbox=777777           ; Mailbox for message waiting indicator