Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
On Tuesday 13 May 2008, Steve Totaro wrote: Can you describe exactly how you are utilizing it, including LAN/WAN, switches, ping times, and other network central details. TDMoE adds the E (ethernet) component to troubleshooting and I think do to this, it may be very fragile depending on network conditions. Don't make the mistake of just focusing on Asterisk and Zaptel in your troubleshooting process. Hi! Thank you very much for your suggestions. Yesterday evening, the telco technician disabled a feature they call NT1 on their side, which is supposed to be a protocol for line monitoring. We tested again and were unable to reproduce the call dropping problem. We established 15 calls to our private extensions to fill all 30 channels and had the calls running throughout the night. I just checked, and they were all still up and running. As it currently appears to me, this line monitoring feature caused the problems. Unfortunately I have been unable to find anything related to NT1 and line monitoring on the internet. I've been in touch with the telco (Telekom Austria) and they will try to find some information concerning this feature. I'll report back with more info as soon as possible. Once again, thank you very much for your support! I really hope this issue is solved (I've been searching for the cause for more than two weeks now). Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
On Tuesday 13 May 2008, Steve Totaro wrote: You can be shot several times and not die. I would try resetinterval=never just to be able to to say Not the problem rather than Probably not the problem. I'll do that, although I'm pretty sure that the setting is not the problem as the yellow alarm occured quite often a few minutes after restarting asterisk and the default is 3600 seconds. PRI debug info would be a great help too. The log I sent in the original message contains pri debug messages. I just had another look at it. Thanks for your help! Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
Florian Hackenberger wrote: On Tuesday 13 May 2008, Steve Totaro wrote: You can be shot several times and not die. I would try resetinterval=never just to be able to to say Not the problem rather than Probably not the problem. I'll do that, although I'm pretty sure that the setting is not the problem as the yellow alarm occured quite often a few minutes after restarting asterisk and the default is 3600 seconds. PRI debug info would be a great help too. The log I sent in the original message contains pri debug messages. I just had another look at it. I did not follow the thread, but can this be a timing problem ? It might be that the far end goes into maint mode due to slips, or what ever. Regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
On Tue, May 13, 2008 at 9:57 AM, Florian Hackenberger [EMAIL PROTECTED] wrote: On Tuesday 13 May 2008, Steve Totaro wrote: You can be shot several times and not die. I would try resetinterval=never just to be able to to say Not the problem rather than Probably not the problem. I'll do that, although I'm pretty sure that the setting is not the problem as the yellow alarm occured quite often a few minutes after restarting asterisk and the default is 3600 seconds. PRI debug info would be a great help too. The log I sent in the original message contains pri debug messages. I just had another look at it. Thanks for your help! Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at Ah, didn't even know you could add attachments to postings to the list. It is a little hard to read and quite a bit of info that may or may not be a problem. Does this happen enough, or do you have enough time to sit there and catch the exact log output when it happens? If you can comment out the spans you are not using, that would reduce a bit of output (I assume you have a single E1 and some POTS (although I don't see them configured, but you said in your initial posting that you could dial out on POTS). ERROR[7968]: chan_zap.c:8176 zt_pri_error: !! Got reject for frame 120, retransmitting frame 120 now, updating n_r! and ERROR[7968]: chan_zap.c:8176 zt_pri_error: !! Got I-frame while link state 2 -- Got UA from network peer Link up. looks suspicious. Maybe Red-fone could give you some insight on these errors. If you can narrow it down then I am sure someone can better. Again, comment out spans not in use, set your verbose to 0, and turn on PRI debugging and try to catch only the event/s that correlate with the calls being dropped. I saw Red-fone's products at Astricon, they looked great for failover. Can you describe exactly how you are utilizing it, including LAN/WAN, switches, ping times, and other network central details. TDMoE adds the E (ethernet) component to troubleshooting and I think do to this, it may be very fragile depending on network conditions. Don't make the mistake of just focusing on Asterisk and Zaptel in your troubleshooting process. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
On Friday 09 May 2008, Steve Totaro wrote: Try resetinterval=never Thank you very much for your suggestion, I'll try the setting and will report back whether it solved our problems. Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
On Friday 09 May 2008, Steve Totaro wrote: Try resetinterval=never Hi! I tried setting resetinterval=60 and established a call. The call survived several channel resets successfully, so this setting is probably not the problem. Any more ideas? Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
On Fri, May 9, 2008 at 5:32 PM, Florian Hackenberger [EMAIL PROTECTED] wrote: Hi! We are using a red-fone bridge (connected to an E1 line from the Telekom Austria) to provide TDMoE connectivity to our asterisk server (ubuntu 7.10, asterisk 1.4.10 from ubuntu, libpri 1.4.0-2, zaptel 1.4.9.2 from http://support.red-fone.com/downloads/zaptel/). We can establish outgoing calls and receive incoming calls from the POTS. Sometimes however (no aparent triggers) all established calls suddenly drop, the span goes into YELLOW alarm and is restarted. Right after the restart we can use the lines as if nothing ever happened. Just sound quality seems to be decreased. The server is a real machine Intel Core 2 with a dedicated 3Com NIC for the red-fone bridge. We have been in touch with red-fone support for hours. They suggested a loopback cable test (which suceeded, we had a few calls up for about 15 hours) and then concluded that either the E1 line must have a problem or it has to be something with asterisk. Today we had a telco technician on-site who told us that the line works as expected and the configuration settings of the redfone bridge (framing, encoding, timing, etc.) are correct. I'm attaching the logfile with span debug information, as well as the relevant snippets from our configuration files. Maybe someone could help us with a few suggestions, as we are a bit lost at this point. Cheers, Florian === zaptel.conf === dynamic=ethmf,eth4/00:50:C2:65:D2:1A/0,31,2 dynamic=ethmf,eth4/00:50:C2:65:D2:1A/1,31,1 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 alaw=1-62 loadzone = us defaultzone=us == === redfone.conf === [globals] fb=192.168.1.254 port=1 server=00:1C:C4:8E:8C:7B [span1] framing=ccs encoding=hdb3 slave crc4 [span2] framing=ccs encoding=hdb3 slave crc4 == === zapata.conf === [trunkgroups] [channels] pridialplan=dynamic prilocaldialplan=local internationalprefix = 00 nationalprefix = 0 localprefix = 03142 usecallerid=yes hidecallerid=no hidecalleridname=no usecallingpres=yes context=default group=0 switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31,32-46,48-62 == -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Try resetinterval=never ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
