In article <4feccd0c.1020...@fivecats.org>, James Sharp <ja...@fivecats.org> wrote: > On 6/28/2012 3:53 PM, Ernie Dunbar wrote: > > 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@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. > > > You've got the spans configured as "group = 63" but you're trying to > dial out on group 3 (DAHDI/g3 rather than DAHDI/g63).
No, the group=63 lines are actually redundant. It is the settings *above* each channel=> line that get applied to the channels when they are created. To the OP: what does "pri show span 3" give you on Voip1? It might be useful to see the complete chan_dahdi.conf from Voip1. To save space, you can list it without comments like this: # grep -v '^;' /etc/asterisk/chan_dahdi.conf Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _____________________________________________________________________ -- 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