Displaying 1 result from an estimated 1 matches for "peoplehere".
2004 Jan 22
3
Grandstream 101
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