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 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
