[asterisk-users] PRI trunk between Asterisk servers does not work.

[Ernie Dunbar]

We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard 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 at local:1] Dial("DAHDI/5-1",  
"DAHDI/g3/604482XXXX") in new stack
[Jun 28 12:43:32] WARNING[4219]: 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 1

I'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.