[asterisk-users] GXP2000 and asterisk 1.0.9

Giordano Grandis g.grandis at invidea.it
Wed Feb 13 09:58:38 CST 2008


Hi all gusy,
i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in "busy" state, if you call it get the busy tone but the phone can male any type of call.
This is my sip.conf
 
[502]
language = it
username = 502
secret = <password>
host = dynamic
type = friend
context = local
canreinvite = yes
dtmfmode = info
callgroup = 1
pickupgroup = 1
callerid = 502 <502>

Under Grandstream's support suggest, I set "Use randmom port" to yes and "Nat traversal (STUN)" to "No, but send keep alive" but without success.
This is the firmware version: Program-- 1.1.5.15    Bootloader-- 1.1.5.6 
 
Anyone can help me ?
 
Thanks in advance
 
Giordano        

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