[Asterisk-Users] Connection Timeout problem with SIP phones from Gnet

[Jean-Francois Theroux]

Hey guys,

	I'm fairly new to Asterisk. Our objective is to have a VoIP PBX 
connected to our PSTN lines. So, right now, I have a box running OpenNA 
Linux, with a 2.4.29 kernel. Asterisk 1.07 and the latest Zaptel drivers 
also.

	I have 2 Gnet SIP phones connected on the same switch as the Asterisk 
box. So far, our phones authenticate with *, because when I do "sip show 
users", I see our 2 phones there.

	The problem I have is this, when I try to dial the other extension, in 
this case 502, from 501, after a few seconds, I get a busy signal. If I 
check on the phone's logs, it says connection timeout.

Here's my dialplan, keep in mind, all the outgoing and incoming stuff is 
irrelevant, since there's no PSTN line connected to it. Only the VoIP 
matters for now.

extensions.conf:
[globals]
JIEF=SIP/501
TEST=SIP/502

[incoming]
exten => s,1,Answer()
exten => s,2,Playback(goodbye)
exten => s,3,Hangup()

[internal]
exten => _5XX,1,Dial(SIP/${EXTEN})
include => outgoing

[outgoing]
ignorepat => 9
exten => _9NXXNXXXXXX,1,Dial(${LOCALTRUNK}/${EXTEN:1})
exten => _9NXXNXXXXXX,2,Playback(invalid)
exten => _9NXXNXXXXXX,3,Hangup

[prompts]
exten => *1,1,Answer()
exten => *1,2,Record(test:gsm)
exten => *1,3,Playback(test)
exten => *1,4,Hangup()

And here's sip.conf:
[general]
port=5060
bindaddr=172.16.1.200
srvlookup=yes
dtfmmode=inband
allow=all

[501]
type=friend
host=172.16.1.201
canreinvite=yes
context=internal
username=501
secret=1234
allow=all
dtfmmode=inband

[502]
type=friend
host=172.16.1.202
canreinvite=yes
context=internal
username=502
secret=1234
allow=all
dtfmmode=inband

Cheers,

-- 
Jean-Francois Theroux
Systems administrator
PrivalODC
450.761.9973
http://www.privalodc.com