Re: [Asterisk-Users] Analog FXO Woes Continue
In addition to the suggestions with the quality of the phone lines, I had the same problem when I had an IRQ conflict with two cards in the same * box. I actually fixed the problem by just swapping the cards between the two slots they were in. Two TDM cards and just flipped them and they both now work. IMHO, just taking the card out and putting the card in another * box only proves the card works. It does not prove the card works properly in a particular card slot on a given * box with a given * configuration/installation. Check your IRQ's and play with them and also play with different card slots in the * box. Lyle - Original Message - From: Paul Dugas [EMAIL PROTECTED] To: Asterisk Mailing List [EMAIL PROTECTED] Sent: Tuesday, December 07, 2004 9:31 AM Subject: [Asterisk-Users] Analog FXO Woes Continue I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog FXO Woes Continue
I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog FXO Woes Continue
[EMAIL PROTECTED] wrote: I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? You should probably scope the lines with a circuit tester. Used Wilcom T136 units can be had on eBay for about 20 bucks. They'll allow you to check the noise and loss on the circuit. When you report it you don't have to describe a problem, but simply state that the circuit is out of spec. No guarantee that this is your problem, but from the symptoms you describe you are definitely on the right track. Good luck. Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog FXO Woes Continue
Hi I feel your pain! We have had the same problem with our telco lines but found that converting to ISDN helped. If the delay on the send and receive two pair is to big the echo canceller is not strong enough. Try using a Voictronix card as they seem to solve the problem to a degree but I would suggest ISDN. Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Dugas Sent: Tuesday, December 07, 2004 5:32 PM To: Asterisk Mailing List Subject: [Asterisk-Users] Analog FXO Woes Continue I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog FXO Woes Continue
I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Don't have any real answers, but might check the following... at least to rule them out. Telco folks _always_ check lines from their demarc (which in some cases is the protector box on the outside of the building). Most will not come inside to measure anything from the customer equipment jack. If that's true in your case, then you have to question the cabling inside the building (to asterisk). That cabling is most often simple inside wire that can easily pick up noise (eg, induction from florescent lights, motors, wall-wart transformers, some desk lamps). If you don't know where the inside wire is run, might try to find out or bypass it with cabling laying on the floor for at least an elementary test. If you did not _see_ a telco person on site doing the transmission checks, you have to assume that someone did them from the central office (most common approach). That's okay in many cases, but its not okay in other more serious cases. The majority of the telco people that would be dispatched for testing only know enough to follow printed procedures using whatever testset they've been given; they don't have the skills to actually interpret the readings for cases they've never seen or been trained to recognize. Its not hard to plug an ordinary phone into the same rj11 jack used by asterisk. Do it and listen close. Given the problems that you've stated, it should not be difficult to hear noise, hum, low volume, etc, if it is in fact bad lines. Also, compare lines; it is not very often four of four lines go bad in exactly the same way. Can you hear any difference between lines? Bridge an ordinary phone on the same pstn line as asterisk. Place some calls from asterisk and listen to what's going on via the analog phone. (Example: some central offices don't like dtmf tones within xxx milliseconds after going off-hook. You'll get wrong numbers, etc. Insert the 'w' option in your Dial statement to delay those dtmf tones a little bit.) To be a little sneaky, unscrew and remove the mouthpiece from the analog phone and you can monitor calls all day long without impacting asterisk's ability to handle calls. If asterisk is having an echo issue (as an example) and you don't hear it with the bridged phone, you at least know where to look. If you messed with the txgain/rxgain for your analog lines, go back to zero gain, use echocancel=yes echotraining=800 rxgain=0.0 txgain=0.0 on each pstn line, reboot the server, and test using some of the above steps to verify problems. If you're still not sure what's going on, transmission test sets are sold by many different companies that you can use from the asterisk rj11 jack to prove line quality. New sets run about $400 to $600 for what you need; check ebay for used pricing. The telco's have a telephone number for a quiet termination and another one for their milliwatt generator. Get those numbers and use the test set to measure noise (quiet termination) and loss (milliwatt generator). If those results are reaonable, then you've got an asterisk configuration problem (and/or digium card problem). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog FXO Woes Continue
We have also decided that FXO interfacing is not reliable enough, even when using the Digium 4 four port FXO card, the lines hang frequently and there have been various quality issues. All of our production deployments are PRI interface, and they are rock solid. While I have not done it myself, it seems that the solution to provide digital interface without the expense of a PRI would be ISDN using a card like the Eicon Diva BRI. No experience with this from our end, but in our region a BRI is less money than two busienss DS0s anyways. From: [EMAIL PROTECTED] on behalf of Rich Adamson Sent: Tue 12/7/2004 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Analog FXO Woes Continue I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Don't have any real answers, but might check the following... at least to rule them out. Telco folks _always_ check lines from their demarc (which in some cases is the protector box on the outside of the building). Most will not come inside to measure anything from the customer equipment jack. If that's true in your case, then you have to question the cabling inside the building (to asterisk). That cabling is most often simple inside wire that can easily pick up noise (eg, induction from florescent lights, motors, wall-wart transformers, some desk lamps). If you don't know where the inside wire is run, might try to find out or bypass it with cabling laying on the floor for at least an elementary test. If you did not _see_ a telco person on site doing the transmission checks, you have to assume that someone did them from the central office (most common approach). That's okay in many cases, but its not okay in other more serious cases. The majority of the telco people that would be dispatched for testing only know enough to follow printed procedures using whatever testset they've been given; they don't have the skills to actually interpret the readings for cases they've never seen or been trained to recognize. Its not hard to plug an ordinary phone into the same rj11 jack used by asterisk. Do it and listen close. Given the problems that you've stated, it should not be difficult to hear noise, hum, low volume, etc, if it is in fact bad lines. Also, compare lines; it is not very often four of four lines go bad in exactly the same way. Can you hear any difference between lines? Bridge an ordinary phone on the same pstn line as asterisk. Place some calls from asterisk and listen to what's going on via the analog phone. (Example: some central offices don't like dtmf tones within xxx milliseconds after going off-hook. You'll get wrong numbers, etc. Insert the 'w' option in your Dial statement to delay those dtmf tones a little bit.) To be a little sneaky, unscrew and remove the mouthpiece from the analog phone and you can monitor calls all day long without impacting asterisk's ability to handle calls. If asterisk is having an echo issue (as an example) and you don't hear it with the bridged phone, you at least know where to look. If you messed with the txgain/rxgain for your analog lines, go back to zero gain, use echocancel=yes echotraining=800 rxgain=0.0 txgain=0.0 on each pstn line, reboot the server, and test using some of the above steps to verify problems. If you're still not sure what's going on, transmission test sets are sold by many different companies that you can use from the asterisk rj11 jack to prove line quality. New sets run about $400 to $600 for what you need; check ebay for used pricing. The telco's have a telephone number for a quiet termination and another one for their milliwatt generator
RE: [Asterisk-Users] Analog FXO Woes Continue
I had a similar problem with DTMF detection using a T100X and TDM400P (1 FXS channel) my analog phone connected to TDM400P would not detect all keys when I tried dialing out. I traced it to a misconfigured extensions.conf file, This was the last file I changed when the problem started. After various attempts to undo my changes (no backup) I deleted the contents of extensions.conf and started over with a basic configuration and restarted * problem disappeared. - Jose -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Dugas Sent: Tuesday, December 07, 2004 7:32 AM To: Asterisk Mailing List Subject: [Asterisk-Users] Analog FXO Woes Continue I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog FXO Woes Continue
[EMAIL PROTECTED] wrote: I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Another thing to find out is whether there are loading coils on the circuit. That can cause all kinds of strange problems if you're not using a purely electromechanical analogue phone. Did you test these circuits using a regular fax machine plugged directly into the circuit? Can you test with a good old-fashioned 56K modem; are you able to connect at a minimum of 28.8K? The fact is that line impairments can be quite expensive for telcos to fix. Right or wrong, it is a part of the reason why they'll attempt to convince you to give up on getting the problem solved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog FXO Woes Continue
[EMAIL PROTECTED] wrote: I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Don't have any real answers, but might check the following... at least to rule them out. Telco folks _always_ check lines from their demarc (which in some cases is the protector box on the outside of the building). Most will not come inside to measure anything from the customer equipment jack. If that's true in your case, then you have to question the cabling inside the building (to asterisk). That cabling is most often simple inside wire that can easily pick up noise (eg, induction from florescent lights, motors, wall-wart transformers, some desk lamps). If you don't know where the inside wire is run, might try to find out or bypass it with cabling laying on the floor for at least an elementary test. Testing from the demarcation point is essential, and poor inside cabling can contribute to the problem, but if the cable is Cat 3 or better, it is unlikely that it will be succeptible to induced noise; that's why twisted pair is twisted - to protect it from induced noise. If you did not _see_ a telco person on site doing the transmission checks, you have to assume that someone did them from the central office (most common approach). That's okay in many cases, but its not okay in other more serious cases. The majority of the telco people that would be dispatched for testing only know enough to follow printed procedures using whatever testset they've been given; they don't have the skills to actually interpret the readings for cases they've never seen or been trained to recognize. Its not hard to plug an ordinary phone into the same rj11 jack used by asterisk. Do it and listen close. Given the problems that you've stated, it should not be difficult to hear noise, hum, low volume, etc, if it is in fact bad lines. Also, compare lines; it is not very often four of four lines go bad in exactly the same way. Can you hear any difference between lines? This is not a bad idea, but is not always conclusive. I've done numerous tests on circuits where it sounded great on a butt set, but was nevertheless out of spec. Also, if the problem is due to loss, it is quite reasonable to expect all the lines to have the exact same problem, because they will all be exactly the same distance from the C.O. Bridge an ordinary phone on the same pstn line as asterisk. Place some calls from asterisk and listen to what's going on via the analog phone. (Example: some central offices don't like dtmf tones within xxx milliseconds after going off-hook. You'll get wrong numbers, etc. Insert the 'w' option in your Dial statement to delay those dtmf tones a little bit.) To be a little sneaky, unscrew and remove the mouthpiece from the analog phone and you can monitor calls all day long without impacting asterisk's ability to handle calls. Say WHAT?!?! OK look, I'm sorry, but this is just plain wrong. Disconnecting the transmitter in your handset will not alter the fact that you have introduced a device in the loop that is in an off-hook condition. To do what you are suggesting, one needs a butt set; which is equipped to passively monitor the line without affecting it. If asterisk is having an echo issue (as an example) and you don't hear it with the bridged phone, you at least know where to look. That isn't really true. Since the analogue phone will not have a transcoding delay, the echo might still be there, just ocurring at the same time as the sidetone. If you messed with the txgain/rxgain for your analog lines, go back to zero gain, use echocancel=yes echotraining=800 rxgain=0.0 txgain=0.0 on each pstn line, reboot the
Re: [Asterisk-Users] Analog FXO Woes Continue
analog phone. (Example: some central offices don't like dtmf tones within xxx milliseconds after going off-hook. You'll get wrong numbers, etc. Insert the 'w' option in your Dial statement to delay those dtmf tones a little bit.) To be a little sneaky, We had one line, it happened to be a business line, that required putting a w before the number in the dial statement in order for it to work. Fixed our problem, Regards Greg Cirino ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog FXO Woes Continue
Don't have any real answers, but might check the following... at least to rule them out. Telco folks _always_ check lines from their demarc (which in some cases is the protector box on the outside of the building). Most will not come inside to measure anything from the customer equipment jack. If that's true in your case, then you have to question the cabling inside the building (to asterisk). That cabling is most often simple inside wire that can easily pick up noise (eg, induction from florescent lights, motors, wall-wart transformers, some desk lamps). If you don't know where the inside wire is run, might try to find out or bypass it with cabling laying on the floor for at least an elementary test. Testing from the demarcation point is essential, and poor inside cabling can contribute to the problem, but if the cable is Cat 3 or better, it is unlikely that it will be succeptible to induced noise; that's why twisted pair is twisted - to protect it from induced noise. Inside wire in the US is NOT twisted pair. That _was_ the point. Doubtful it is in canada either. Check any of the cable specs for the 4-wire el-cheapo inside wire that's been in use for years. Inside wire has been known to create issues for well over twenty years _if_ the cable is located anywhere near noise-generating electrical devices. If you did not _see_ a telco person on site doing the transmission checks, you have to assume that someone did them from the central office (most common approach). That's okay in many cases, but its not okay in other more serious cases. The majority of the telco people that would be dispatched for testing only know enough to follow printed procedures using whatever testset they've been given; they don't have the skills to actually interpret the readings for cases they've never seen or been trained to recognize. Its not hard to plug an ordinary phone into the same rj11 jack used by asterisk. Do it and listen close. Given the problems that you've stated, it should not be difficult to hear noise, hum, low volume, etc, if it is in fact bad lines. Also, compare lines; it is not very often four of four lines go bad in exactly the same way. Can you hear any difference between lines? This is not a bad idea, but is not always conclusive. I've done numerous tests on circuits where it sounded great on a butt set, but was nevertheless out of spec. Also, if the problem is due to loss, it is quite reasonable to expect all the lines to have the exact same problem, because they will all be exactly the same distance from the C.O. The point was the poster is suggesting some very serious line deficiencies, and if those deficiencies are truly the result of bad lines, he should be able to detect at least _some_ issues by using at least some of his five senses. Bridge an ordinary phone on the same pstn line as asterisk. Place some calls from asterisk and listen to what's going on via the analog phone. (Example: some central offices don't like dtmf tones within xxx milliseconds after going off-hook. You'll get wrong numbers, etc. Insert the 'w' option in your Dial statement to delay those dtmf tones a little bit.) To be a little sneaky, unscrew and remove the mouthpiece from the analog phone and you can monitor calls all day long without impacting asterisk's ability to handle calls. Say WHAT?!?! OK look, I'm sorry, but this is just plain wrong. Disconnecting the transmitter in your handset will not alter the fact that you have introduced a device in the loop that is in an off-hook condition. Better try it before you knock it (but use a real analog set, not the el-cheapo electronic ones). Disconnecting the mic is exactly the same thing as the old multi-party phones with the little switch on its side. (In fact, playing with the mic use to be one way to bypass coin operated requirements. :) To do what you are suggesting, one needs a butt set; which is equipped to passively monitor the line without affecting it. Better take your butt set apart, draw the schematic, and do the same for what is stated above. If asterisk is having an echo issue (as an example) and you don't hear it with the bridged phone, you at least know where to look. That isn't really true. Since the analogue phone will not have a transcoding delay, the echo might still be there, just ocurring at the same time as the sidetone. If you actually think about what you just said, you'll probably want to take that statement back. Think real hard though! (Oh well, let me give you a clue: near-end verses far-end.) The telco's have a telephone number for a quiet termination and another one for their milliwatt generator. Get those numbers and use the test set to measure noise (quiet termination) and loss (milliwatt generator). If those results are reaonable, then you've got an asterisk configuration problem (and/or digium card problem).
Subject: Re: [Asterisk-Users] Analog FXO Woes Continue
On Tue, 2004-12-07 at 09:34 -0600, asterisk-users- [EMAIL PROTECTED] wrote: I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. Hi, some quick suggestions may get you results. a) make an appointment with the most helpful person in that local telco switch (anybody!) who has the expertise to help you with analog telephone OUTSIDE CABLE PLANT problems. Tell him you suspect cross talk, noise, grounding or mis-wire of the pairs. b) announce system maintenance time if your main trunks are all analog, if not, simply disable those extensions by RENUMBERING them from your exten.conf, so that regular users don't have access to them. ONLY YOU AND YOUR TEST ENGINEER FROM THE CO should test these lines. c) Please have your hearing checked (you will need this feature) or get a good telephone butt test set, so that you can hear the audio quality clearly and I want to say that you must be able to determine if you are getting cross-talk from another line, which normally you would not hear when you are talking to someone. If you are getting noise / cross-talk, mis-wiring (A1+B2 and A2+B1 of two telephone pairs, A, B) you will get mixed results. d) If you can, physically uproot and disconnect the main lines from your incoming demarc and temporarily replace those wires with a long spool of normal cat-5 cables (the UTP Ethernet kind..) doesn't matter, but make sure the wires are SECURELY fastened at the joints so you have no loose connections. What I mean is if you had... d.1) [diagram] telco -- demarc/wall junction box/internal wiring/asterisk change it to d.2) [diagram] telco -- demarc/wall junction box/very long spool of wire/RJ-11 block/asterisk e) Note that on your Asterisk FXO ports, typically there is RJ-45, instead of RJ-11 connectors, so you will have to fashion a RJ-45 plug with only the MIDDLE two wires connected and then terminate the other end of the flat cable with a RJ-11 cable. It doesn't hurt to make this cable very long, as long as you don't trip over it. f) Please make sure you study up and understand the implications of TXGAIN and RXGAIN and how to use it, I recently screwed up my system by INCREASING the gain, and obviously increasing the analog noise channel, so you may want to DECREASE gain by adjusting the RX parameter. Don't play with the TX parameter just yet, only test ONE at a time. Have the external engineer call from various places, inside your office (from you), outside your office (to asterisk) and make sure you see the console real time messages in highly verbose mode so you can trap the error if it happens. WHAT IS THE ERROR MESSAGE from Zaptel if any ? g) Hook up a test set IN PARALLEL to the incoming line at the demarc, and THEN SEPARATELY at the asterisk location.Make a habit of when the engineer from outside (it could actually be one of your colleagues!!) calls in, you SPY and MONITOR the audio level / quality of your connection, and if you suspect a problem, (it should sound CRYSTAL CLEAR and the SAME LEVEL) you can divert the incoming line into your test set and then have them call you, and you be the judge of what the level is. h) If you are aware of any humidity problems now is the time to speak up, as if a telephone pair wire is near a humid / wet location/pipe/gutter/roof/floor, it will provide the equivalent of audio-limited-bandwidth service, and most of your DTMF signals and audio conversations will be affected. i) If you can, please use TWISTED pair wires for the internal connections (in-house) and also consider strongly testing the whole setup in reverse direction by CALLING OUT and asking the other side how do you hear me ? j) Also please note that you need to ask your engineers what average DC voltages you should be expecting for off-hook conditions (use a DC 48Volts meter and the excellent guide available on the Wiki page), and also http://www.teracomtraining.com/tutorials/teracom-tutorial-PSTN.htm, and also the expected ring voltage when CO rings the FXO port. Please make sure you know the names of the points at which you are testing, so that you can be knowledgeable and provide that input to the engineer who will then treat you quite well. If you can tell them to test the line upto your asterisk premise, they may accomodate you just this once, and that will help you incredibly. k) If you did get satisified, they would get revenue, OR THEY WILL LOSE YOU, that is their motiviation. Finally, analog telephone lines are simpler to fix than digital, only because they can be diagnosed within
RE: [Asterisk-Users] Analog FXO Woes Continue
Rich Adamson wrote: Don't have any real answers, but might check the following... at least to rule them out. Telco folks _always_ check lines from their demarc (which in some cases is the protector box on the outside of the building). Most will not come inside to measure anything from the customer equipment jack. If that's true in your case, then you have to question the cabling inside the building (to asterisk). That cabling is most often simple inside wire that can easily pick up noise (eg, induction from florescent lights, motors, wall-wart transformers, some desk lamps). If you don't know where the inside wire is run, might try to find out or bypass it with cabling laying on the floor for at least an elementary test. Testing from the demarcation point is essential, and poor inside cabling can contribute to the problem, but if the cable is Cat 3 or better, it is unlikely that it will be succeptible to induced noise; that's why twisted pair is twisted - to protect it from induced noise. Inside wire in the US is NOT twisted pair. That _was_ the point. Doubtful it is in canada either. Check any of the cable specs for the 4-wire el-cheapo inside wire that's been in use for years. Inside wire has been known to create issues for well over twenty years _if_ the cable is located anywhere near noise-generating electrical devices. LOL! Inside wiring used to be untwisted, many years ago. But UTP has been in use for so long now it'd be very unusual to find a building without it. In fact, I have never seen anything less than CAT2, and I have been involved in a lot of PBX installations; many of them in very old buildings indeed. You are correct that untwisted cabling will be susceptible to the aforementioned problems. You are incorrect in asserting that such cable is common. Perhaps you are talking about the grey cords with RJ11s on the ends that run from the wall to your phone. Those are not typically twisted, but they are also NOT referred to as inside wiring. Those should not be used to terminate telco circuits, and, if so used, should not be more than six feet in length. Also, you would never run those through walls or plenum, not only for technical reasons, but also because they would generally violate fire code. If you did not _see_ a telco person on site doing the transmission checks, you have to assume that someone did them from the central office (most common approach). That's okay in many cases, but its not okay in other more serious cases. The majority of the telco people that would be dispatched for testing only know enough to follow printed procedures using whatever testset they've been given; they don't have the skills to actually interpret the readings for cases they've never seen or been trained to recognize. Its not hard to plug an ordinary phone into the same rj11 jack used by asterisk. Do it and listen close. Given the problems that you've stated, it should not be difficult to hear noise, hum, low volume, etc, if it is in fact bad lines. Also, compare lines; it is not very often four of four lines go bad in exactly the same way. Can you hear any difference between lines? This is not a bad idea, but is not always conclusive. I've done numerous tests on circuits where it sounded great on a butt set, but was nevertheless out of spec. Also, if the problem is due to loss, it is quite reasonable to expect all the lines to have the exact same problem, because they will all be exactly the same distance from the C.O. The point was the poster is suggesting some very serious line deficiencies, and if those deficiencies are truly the result of bad lines, he should be able to detect at least _some_ issues by using at least some of his five senses. Do you say this from experience? Because I _have_ seen lines that sounded perfect with a butt set, and nevertheless measured out of spec. My _experiences_ do not support your _theory_. Sure, he can test the line as you suggest, and if he detects noise he can report it as such. If, however, nothing can be detected, it does NOT indicate that the circuit is nominal. Bridge an ordinary phone on the same pstn line as asterisk. Place some calls from asterisk and listen to what's going on via the analog phone. (Example: some central offices don't like dtmf tones within xxx milliseconds after going off-hook. You'll get wrong numbers, etc. Insert the 'w' option in your Dial statement to delay those dtmf tones a little bit.) To be a little sneaky, unscrew and remove the mouthpiece from the analog phone and you can monitor calls all day long without impacting asterisk's ability to handle calls. Say WHAT?!?! OK look, I'm sorry, but this is just plain wrong. Disconnecting the transmitter in your handset will not alter the fact that you have introduced a device in the loop that is in an off-hook condition. Better try it before you knock it (but use a real analog set, not the el-cheapo
