On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar <[email protected]> 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. > > ---------------------------------------------------------------- > 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
Hello Ernie, Could you post the dahdi/system.conf from both voip1 and voip3 servers? I suspect that you have not correctly defined the data channel (dchan setup should be in system.conf and not in chan_dahdi.conf, where I see a not necessarily dchannel configuration) HTH, Ioan -- _____________________________________________________________________ -- 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
