We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has aWildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server.
We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however.
At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem.
- I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3.
- I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot.- DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same.
The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" group=3 context=default switchtype = national signalling = pri_net channel => 49-71 group = 63 ; Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" group=4 context=default switchtype = national signalling = pri_net channel => 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=>1-23 dchannel=>24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten => 604484XXXX,1,Dial(DAHDI/g3/604482YYYY)But when I call 604484XXXX from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1:
-- Executing [604484XXXX@local:1] Dial("DAHDI/5-1", "DAHDI/g3/604482XXXX") in new stack [Jun 28 12:43:32] WARNING: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839ZZZZ' to '604484XXXX' on channel 0/5, span 1I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly.
Any input on this problem is greatly appreciated. ---------------------------------------------------------------- This message was sent using Lightspeed.ca's Advanced Webmail. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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