[Asterisk-Users] Grandstream 101

[Siggi Langauf]

On Thu, 22 Jan 2004, 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.
> [gs]
> canreinvite=no
> dtmfmode=info

To solve 1., use dtmfmode = rfc2833 or just leave it empty.

> In the GS101 setting
> rtp port = 5004
> sip port = 5060
> dtmf = sip info

using "via RTP (RFC2833)" here works fine for me.

> codec = pcmu
> codec = pcma
>
> Any pointer of a sample of config file would be most appreciate.

WRT the codecs, Setting all 6 choices in the grandstream web interface has
helped for me, most of the time.
One phone required several reset cycles before it would accept new
settings, though. Another one only accepted the new settings after
unplugging/replugging the power supply. This one also lost its settings
during another power supply.

I guess these "phones" are just a bit flakey WRT their settings...

HTH,
	Siggi