[asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco <-cA-> Machine1 <-cB-> Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1 Cable Configuration files are included at the end of this message. I have used both 'cat /proc/interupts' and 'lspci -vb' to verify that the cards do not have IRQ conflicts. Machine2 can be plugged into the telco pri and works fine. Machine1 works on the telco PRI so long as I have removed the configuration for the second span (the one on the crossover). I can leave the card for the second span in - so long as it is not configured. Machine 1 CLI Notices If both cards are configured and connected. Zttool reliably reports no alarms. Asterisk appears to start without errors at verbosity three. Very quickly after it starts Asterisk starts reporting WARNING[28899] chan_zap.c:6668 handle_init_event: Detected alarm on channel NN: Red Alarm WARNING[28899] chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel NN It will output these for all the channels on the second span (NN ranges from 25 to 47). It then will report NOTICE[28898] chan_zap.c:8460 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 2 WARNING[28898] chan_zap.c:2393 pri_find_dchan: No D-channels available! Using primary channel 48 as D-channel anyway! This is immediately followed by a notice that the alarms were cleared on each of the channels of span 2, including the data-channel. NOTICE [29899]: chan_zap.c:6661 handle_init_event: Alarm cleared on channel NN NOTICE [29898]: chan_zap.c:8460 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 2 This is then followed by the first error ERROR [29898]: chan_zap.c:8174 zt_pri_error: !! Got S-frame while link down I then get constant notices that "Primary D-Channel on span 2 up". This will be periodically broken up by a repeat of the warnings, notices and errors I describe above. However, now the problems occur for all of the channels of both spans. Machine 1 CLI Notices Machine 1 experiences almost the same behavior on its span. The only differance I am noticing is that instead of the S-frame error I get the following notice: chan_zap.c:8457 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Zap Restart Fails On either machine if I attempt a 'zap restart' I receive WARNING[30686]: chan_zap.c:903 zt_open: Unable to specify channel 1: Device or resource busy ERROR[30686]: chan_zap.c:7160 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp->channel = 1, channel = 1 ERROR[30686]: chan_zap.c:10467 build_channels: Unable to register channel '1-23' WARNING[30686]: chan_zap.c:9764 zap_restart: Reload channels from zap config failed! I have not attempted to connect two Asterisk boxes though a T1 crossover before so I am stumped. I have included my zaptel.conf and zapata.conf files below. I will certainly appreciate any help you can give. Thanks, -Brian Machine1 = zaptel.conf --- defaultzone=us loadzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 zapata.conf --- [trunkgroups] [channels] group=1 context=fromtelco signalling=pri_cpe switchtype=national channel=>1-23 group=1 context=frommachine2 signalling=pri_net switchtype=national channel=>25-47 Machine2 = zaptel.conf --- defaultzone=us loadzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] group=1 context=frommachine1 signalling=pri_cpe switchtype=national channel=>1-23 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
Thanks for all of the feedback. I really appreciate the help! :) My cable is 1--4 2--5 4--1 5--2 I got this from http://www.voip-info.org/wiki/view/crossover+T1+cable http://help.fonality.com/index.php/T1_crossover_HOWTO http://www.ldfacts.com/faq_files/How-to-Make-a-T1-Crossover-cable.htm Each span is on a TE120P On Machine1 I have timing set to one on the connection to the telco to use its timing. I have the timing on the second span of Machine1 to zero to provide timing for for Machine2. Machine2 has timing set to one to use the timing from Machine1. -Brian On 9/21/07, Doug Lytle <[EMAIL PROTECTED]> wrote: > > What does your cable pin in/out look like. I haven't connected two > Asterisk systems, but I do have a dual-PRI on one system with a > crossover cable; both spans come up fine. My pin in/out is: > > 1 -- 5 > 2 -- 4 > 4 -- 2 > 5 -- 1 > ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
On 9/20/07, Jared Smith <[EMAIL PROTECTED]> wrote: > > I'd look at your wiring, as an HDLC error like that is usually an > indication of some type of a problem at the physical layer. > I tried two other cables this morning with the same results... While I was swapping cables around I unloaded asterisk and the zaptel modules. One thing I noticed once everything was hooked up and restarted is that Machine2 shows a red alarm but Machine1 shows both spans as clear of alarms... Otherwise everything looks the same as yesterday. Have any of you connected two asterisk machines by t1 crossover using pri_net/pri_cpe signaling? I am completely stumped and would love to know that some had done this and what their configuration look like. Thanks again, -Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
On 9/21/07, Forrest Beck <[EMAIL PROTECTED]> wrote: > > You should also try making/obtaining a T1 loopback adapter and test both > cards. > > http://www.adtran.com/adtranpx/Doc/0/FMPI9MBKGJBH39S8038BE81ID8/CU-2abf1b83844b11d78ff20c045003.html > I have not tested with a loop back but I did verify that each of the cards appears to work when it is used on its own and is connected to the t1 to the telco. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
At this point I do not think the problem is the wiring. What else should I try? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
On 9/24/07, Doug Lytle <[EMAIL PROTECTED]> wrote: > > Have you confirmed that the failing card is working correctly? Maybe > the card is at fault. > All of the cards have been confirmed to work by themselves. -Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
On 9/24/07, Doug Lytle <[EMAIL PROTECTED]> wrote: > > The only other suggestion I have would have would be to use IAX instead > of PRI for inter-machine communications. > LOL Yeah, normally that is what I would use. Unfortunately it is not an option for this... ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
Okay. I ordered a commercially made T1 crossover cable, connected all of the cables and rebooted both computers. I no longer get the 'Got S-frame while link down' or 'HDLC Bad FCS' errors. However, I still receive the series of 'Detected alarm on channel NN: Red Alarm' and 'Unable to disable echo cancellation on channel NN'. Followed by the alarms clearing. At this point I am confident that the cabling is not the problem. I have included my zaptel and zapata conf files again below. Is there a way to make the second TE120P card pass on the timing received from the first? (rather than using software timing for the pri_net signalling) The errors all seem to be about echo cancellation... What do I need to do to force asterisk to never disable echo cancellation? Thanks again for all of the help. Even though we have not found a solution yet I appreciate the help - and am still confident we will succeed! -Brian Machine1 = zaptel.conf --- defaultzone=us loadzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 zapata.conf --- [trunkgroups] [channels] group=1 context=fromtelco signalling=pri_cpe switchtype=national channel=>1-23 group=1 context=frommachine2 signalling=pri_net switchtype=national channel=>25-47 Machine2 = zaptel.conf --- defaultzone=us loadzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] group=1 context=frommachine1 signalling=pri_cpe switchtype=national channel=>1-23 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] One Way Delay in Audio Over Analog
I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service. The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior. The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen). Turning on or off echo cancellation and/or call progress does not seem to change the behavior. I will appreciate any ideas you have. I am certainly stumped. Thanks and Happy New Year! -Brian ___ --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] One Way Delay in Audio Over Analog
Thanks for the replies! The analog card is a TDM400P. The system is currently using asterisk-1.4 r81383 and zaptel-1.4 r2649. I am told that the problem did not exist when the system was using asterisk-1.2.10 and zaptel 1.2.9.1. -Brian On 12/31/07, Brian Alexander <[EMAIL PROTECTED]> wrote: > > I have been trying to track down the cause/fix for a problem and I am out > of ideas... I am hoping one of you can point me in the right direction. > > The symptom is that when a calls is placed from an internal extension > through an analog line to a number on the pstn the caller can hear the > callee but the callee can not hear the caller for as long as ten seconds. > > The problem appears to happen fairly consistently on the same pstn > numbers. However, I have not seen a common characteristic in those numbers. > For example, one of them is a direct number to a cell phone and another is > to a Verizon fiber-optic phone/data service. > > The problem does not seem to be related to the type of SIP phone being > used by the caller - for example, we have tried both X-Lite and Polycom > phones without a change in behavior. > > The problem does not appear to occur if the callee then calls into our > system (at least the one time I was able to have this happen). > > Turning on or off echo cancellation and/or call progress does not seem to > change the behavior. > > I will appreciate any ideas you have. I am certainly stumped. > > Thanks and Happy New Year! > -Brian > ___ --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
[asterisk-users] Problems With Playback of Audio On SIP Only System
I have been installing Asterisk as a SIP only system (no Digium Hardware) for demonstration purposes. SIP users can connect to menus and voicemail fine but the audio quality is terrible. The stock voicemail problems are bad but basically understandable - voice menus recorded through the asterisk-gui-2.0 are difficult to even understand. The phone I am testing with is a Polycom SountPoint IP 430 SIP. I have configured the phone for ulaw to be it primary codec and set disallow all and allow ulaw in the users.conf. When that did not work I guessed that something was wrong with dahdi_dummy but dahdi_test is showing results around 99.987%. Here are the details of what software I have been using: asterisk-1.4 (r168975) dahdi-linux-complete 2.1.0 (r 5662) asterisk-gui-2.0 (r4446) The linux kernel is 2.6.24.6 built with 1000 Hz timer. Thank you for your help, I am a stumped. -Brian ___ -- 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] Problems With Playback of Audio On SIP Only System
Mark, Thanks - that was the problem I was having. Is there somewhere I could have looked to have discovered the problem on my own? I would never have guessed that on my own and my searches had not found it either. Thanks again, -Brian On Mon, Jan 19, 2009 at 7:00 PM, Mark Michelson wrote: > Brian Alexander wrote: > > I have been installing Asterisk as a SIP only system (no Digium > > Hardware) for demonstration purposes. SIP users can connect to menus and > > voicemail fine but the audio quality is terrible. The stock voicemail > > problems are bad but basically understandable - voice menus recorded > > through the asterisk-gui-2.0 are difficult to even understand. > > > > The phone I am testing with is a Polycom SountPoint IP 430 SIP. I have > > configured the phone for ulaw to be it primary codec and set disallow > > all and allow ulaw in the users.conf. > > > > When that did not work I guessed that something was wrong with > > dahdi_dummy but dahdi_test is showing results around 99.987%. > > > > Here are the details of what software I have been using: > > asterisk-1.4 (r168975) > > dahdi-linux-complete 2.1.0 (r 5662) > > asterisk-gui-2.0 (r4446) > > > > The linux kernel is 2.6.24.6 built with 1000 Hz timer. > > > > Thank you for your help, I am a stumped. > > > > -Brian > > If you are using gsm prompts and gcc version 4.2 or higher, then you may be > experiencing the optimizer bug that gcc has with gsm audio. The workarounds > for > this are to use a different format for sounds or to set the DONT_OPTIMIZE > flag > in menuselect. If you want an optimized build and gsm formatted sounds, > then you > could always attempt downgrading your gcc version to 4.1 or earlier. > > Mark Michelson > > ___ > -- 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 > ___ -- 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] Problems With Playback of Audio On SIP Only System
> > Mark is correct - I fruitlessly searched for solutions to audio problems. Is there a diagnostic tool or cli command that might have pointed me in the right direction? When something like this comes up I like to learn the new trick that would have helped. (BTW: I like dahdi_test - thanks to whoever wrote that) Maybe the trick was to ask for help sooner ;-) However, I dislike posting a question until I really know I need help. I appreciate Kevin's going to bat for making it easier to find in the build process. I'm not an Asterisk expert but I've installed half a dozen custom systems now and am at least not a novice anymore. I still don't think I would have found the problem without Mark's help. Thanks again, -Brian ___ -- 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
[asterisk-users] Voice Mail Delete Notification
Do you know if there is a way to have an script run whenever a user has deleted a voicemail message? I want to have multiple users, all with there own passwords, share the same mailbox. When any one of them deletes a message I want it deleted from everyone's mailbox. I believe that I can do this by using symbolic links on the file system for their voice mail folders. However, I would like to keep a record of which users deleted messages when... I do not see a way to hook into the voice mail to do this. Thanks, -Brian ___ -- 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
