Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
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 = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) 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 '778839' to '604484' 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
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
Quoting Ioan Indreias indre...@gmail.com: On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar maill...@lightspeed.ca 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 = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) 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 '778839' to '604484' 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 Okay, here's /etc/dahdi/system.conf (it's unmodified from the autogenerated file): # Autogenerated by /usr/sbin/dahdi_genconf on Mon Jul 26 22:53:04 2010 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,0,esf,b8zs # termtype: te bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 span=3,0,0,esf,b8zs # termtype: te bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 span=4,0,0,esf,b8zs # termtype: te bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global data loadzone = us defaultzone = us This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
Quoting Tony Mountifield t...@softins.co.uk: 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 = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) 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 '778839' to '604484' 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 looks like this: # asterisk -rx 'pri show span 3' Primary D-channel: 72 Status: Provisioned, Down, Active Switchtype: National ISDN Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Logical Channel Mapping: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 The only differences I see between 'pri show span 3' and 'pri show span 4' are that the status on span 4 is Provisioned, Up, Active and that the D-channel is different, which is to be expected. 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 Okay, here you go: [channels] usecallerid=yes cidsignalling=bell cidstart=polarity facilityenable=yes hidecallerid=no callwaitingcallerid=yes callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=no immediate=no group=1 signalling=pri_cpe switchtype=national pridialplan=unknown relaxdtmf=yes context=local channel=1-23 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=3600 #include dahdi-channels.conf And dahdi-channels.conf looks like: group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group=4 context=default switchtype = national signalling = pri_net channel = 73-95
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
Quoting Tony Mountifield t...@softins.co.uk: 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 = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) 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 '778839' to '604484' 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? Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. 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
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
- Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) --Tim -- _ -- 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
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
On 12-06-29 11:40 AM, Tim Nelson wrote: - Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) *CLI pri show version -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
Quoting Tim Nelson tnel...@rockbochs.com: - Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) Excellent! Funny thing about that. Our original plan was to use a SIP trunk until we discovered that faxes don't work worth a damn that way. Ergo, I didn't compile libpri first. 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
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
- Original Message - Quoting Tim Nelson tnel...@rockbochs.com: - Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) Excellent! Funny thing about that. Our original plan was to use a SIP trunk until we discovered that faxes don't work worth a damn that way. Ergo, I didn't compile libpri first. Yep, that'd cause what you're seeing. Glad we could help. :) --Tim -- _ -- 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
[asterisk-users] PRI trunk between Asterisk servers does not work.
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 = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) 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 '778839' to '604484' 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
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar maill...@lightspeed.ca 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 = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) 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 '778839' to '604484' 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
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
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 = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) 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 '778839' to '604484' 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). -- _ -- 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
