[asterisk-users] Phantom rings after FXO/FXS setup
I have recently setup Trixbox 2.6.1 on a machine and configured it with an FXO and FXS module. I can make and receive calls just fine so there is no problem with the configuration of how the ports are set. The problem I am having is when I miss a call. The phone will ring 15 minutes later and continue to ring exactly 15 minutes after that and 15 after that...etc. I cannot find anything online that tells me how to get it to quit this. Any help is greatly appreciated. Thanks. -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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] Phantom rings after FXO/FXS setup
Voicemail indication on the FXS port? I you have voicemail configured the ring is indicating that the extension has a message waiting. On Thu, 2011-09-01 at 14:51 -0500, Chris Ramirez wrote: I have recently setup Trixbox 2.6.1 on a machine and configured it with an FXO and FXS module. I can make and receive calls just fine so there is no problem with the configuration of how the ports are set. The problem I am having is when I miss a call. The phone will ring 15 minutes later and continue to ring exactly 15 minutes after that and 15 after that...etc. I cannot find anything online that tells me how to get it to quit this. Any help is greatly appreciated. Thanks. -- Chris Ramirez TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Phantom rings after FXO/FXS setup
I have already checked that. The voice mail was disabled when it first occurred. So I set the voice mail up and it still happens but with no new messages. On 9/1/2011 3:33 PM, Carlos Chavez wrote: Voicemail indication on the FXS port? I you have voicemail configured the ring is indicating that the extension has a message waiting. On Thu, 2011-09-01 at 14:51 -0500, Chris Ramirez wrote: I have recently setup Trixbox 2.6.1 on a machine and configured it with an FXO and FXS module. I can make and receive calls just fine so there is no problem with the configuration of how the ports are set. The problem I am having is when I miss a call. The phone will ring 15 minutes later and continue to ring exactly 15 minutes after that and 15 after that...etc. I cannot find anything online that tells me how to get it to quit this. Any help is greatly appreciated. Thanks. -- Chris Ramirez TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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 -- _ -- 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 -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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] Phantom Rings
I'm fairly certain the problem is with the phone line. I have all callerID settings disabled as the Telco is unable to provide it along with our rollover line setup due to limitations in their antiquated switch. The CLI and Logs all plainly show the calls as if they were normal calls with the exception of a message about Failed to write frame and no DTMF attempts, then the call is routed into the operator queue. The calls always came in on Zap1-1 so I tried swapping the 2 lines to see if it stayed on port 1 or if the phantom followed the line. As expected, the phantom rings followed the line and began showing up on Zap2-1. So it pretty has to be something in the telco, but I'm not sure what. Putting WaitForRing(3) before the Answer command in my IVR menu eliminates most of them, but sometimes more of them slip through. I get a similar problem with a domestic analogue line in the UK. I *speculate* that there is a short half ring being sent for some reason (line test or similar), but my card (Digium) seems to need about 5 seconds to detect hangup on the remote end, so I get a phantom 2 rings at my end and then it stops... No solution, but thought it might give you something to consider... Ed W ___ -- 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] Phantom Rings
Quoting Ed W [EMAIL PROTECTED]: one thing I thought about, but never actually did was to install a damping circuit across the line - a phone plugged in the line never actually rang or if it did it was so short it was imperceptible. I figured just a load across the might damp down the test pulse enough to not be tricked into ringing the channel bank. social engineering rather than technical engineering eventually solved the problem though. I'm fairly certain the problem is with the phone line. I have all callerID settings disabled as the Telco is unable to provide it along with our rollover line setup due to limitations in their antiquated switch. The CLI and Logs all plainly show the calls as if they were normal calls with the exception of a message about Failed to write frame and no DTMF attempts, then the call is routed into the operator queue. The calls always came in on Zap1-1 so I tried swapping the 2 lines to see if it stayed on port 1 or if the phantom followed the line. As expected, the phantom rings followed the line and began showing up on Zap2-1. So it pretty has to be something in the telco, but I'm not sure what. Putting WaitForRing(3) before the Answer command in my IVR menu eliminates most of them, but sometimes more of them slip through. I get a similar problem with a domestic analogue line in the UK. I *speculate* that there is a short half ring being sent for some reason (line test or similar), but my card (Digium) seems to need about 5 seconds to detect hangup on the remote end, so I get a phantom 2 rings at my end and then it stops... No solution, but thought it might give you something to consider... Ed W ___ -- 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ -- 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] Phantom Rings
I'm having a major problem at one of my branch offices with Phantom Rings on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and have made several calls to Verizon with no solution to the problem. I know the telco switch that is feeding my analog lines is an old switch and can't even do CallerID with 2 lines in a rollover configuration. Audio quality on the line is perfect during voice calls. No static or other noise. I've asked for disconnect supervision to be added to the line, but It doesn't look like it's there. The line still seems to keep the channel open long after the far end hangs up. Has anyone else ever seen this problem or have any ideas how to eliminate it? Thanks, Brent Davidson ___ -- 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] Phantom Rings
Quoting Brent Davidson [EMAIL PROTECTED]: I'm having a major problem at one of my branch offices with Phantom Rings on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and have made several calls to Verizon with no solution to the problem. I know the telco switch that is feeding my analog lines is an old switch and can't even do CallerID with 2 lines in a rollover configuration. Audio quality on the line is perfect during voice calls. No static or other noise. I've asked for disconnect supervision to be added to the line, but It doesn't look like it's there. The line still seems to keep the channel open long after the far end hangs up. Has anyone else ever seen this problem or have any ideas how to eliminate it? I had the phantom rings for years, once a day same time roughly every day, finally just got annoyed enough one day I trapped the telco on the phone with me till I finally got to talk to the right person. The right person knew instantly what I was talking about after months of previous denials. On DMS switches what you need to insist be added to your customer line profile is something called NLT or no line test. The wrong person can even look it up if you tell them the name of it - imagine that eh ? Thanks, Brent Davidson ___ -- 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ -- 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] Phantom Rings
Jon Pounder wrote: I had the phantom rings for years, once a day same time roughly every day, finally just got annoyed enough one day I trapped the telco on the phone with me till I finally got to talk to the right person. The right person knew instantly what I was talking about after months of previous denials. On DMS switches what you need to insist be added to your customer line profile is something called NLT or no line test. The wrong person can even look it up if you tell them the name of it - imagine that eh ? Unfortunately the tech I spoke to said the switch we're connected to is so old it has no built-in test capabilities. To run any sort of line tests the line has to be disconnected from the switch and connected to an external test set. I guess that's one of the things you deal with when the boss decides to serve only rural markets. It would be hard to find locations any more rural than where our branches are. At present I have ztmonitor streaming all line activity to a file. I've got plenty of hard drive space so I can record all day if I need to. According to the manager of the branch in question, there were at least 50 phantom calls yesterday, but there has only been 1 this morning. The other curiosity here is that reviewing my asterisk logs all of the phantom calls are on 1 line and swapping ports, the calls follow the line. It's easy to spot the phantom calls in the logs because they always mention dropped frames (probably because of the dialtone coming from the Analog line card). Thanks, Brent ___ -- 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] Phantom Rings
Quoting Brent Davidson [EMAIL PROTECTED]: Jon Pounder wrote: I had the phantom rings for years, once a day same time roughly every day, finally just got annoyed enough one day I trapped the telco on the phone with me till I finally got to talk to the right person. The right person knew instantly what I was talking about after months of previous denials. On DMS switches what you need to insist be added to your customer line profile is something called NLT or no line test. The wrong person can even look it up if you tell them the name of it - imagine that eh ? Unfortunately the tech I spoke to said the switch we're connected to is so old it has no built-in test capabilities. To run any sort of line tests the line has to be disconnected from the switch and connected to an external test set. I guess that's one of the things you deal with when the boss decides to serve only rural markets. It would be hard to find locations any more rural than where our branches are. At present I have ztmonitor streaming all line activity to a file. I've got plenty of hard drive space so I can record all day if I need to. According to the manager of the branch in question, there were at least 50 phantom calls yesterday, but there has only been 1 this morning. The other curiosity here is that reviewing my asterisk logs all of the phantom calls are on 1 line and swapping ports, the calls follow the line. It's easy to spot the phantom calls in the logs because they always mention dropped frames (probably because of the dialtone coming from the Analog line card). is that one line in the same cable bundle all the way back to the CO ? Could be picking up interference or something, or be half connected to some other line somewhere or some other weirdness. ask for a TDR reading of cable feet on a working and non-working line and they should be damn close to identical. Could also be a bridge tap picking up some stray signal somehow - have that checked for and disconnected. I have also had issues before where an analog modem just plain would not work as the management cct on a telco supplied T1, they changed every piece of wire all the way back to the CO and finally decided it was a bad switch port, swapped the line to a new port and voila worked - tech figured the port was just flaky from lightning damage or something, so some other poor schlub will get a line on there at some point since it tests out ok but doesn't work. I would like to see them go to this effort if it was my own line with my modem - never would have happened. initially all the fingers were pointed at bad building wiring, but that was ruled out in the first 5min, until eventually all that was left was the switch port. Thanks, Brent ___ -- 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ -- 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] Phantom Rings
Brent Davidson wrote: I'm having a major problem at one of my branch offices with Phantom Rings on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and have made several calls to Verizon with no solution to the problem. I know the telco switch that is feeding my analog lines is an old switch and can't even do CallerID with 2 lines in a rollover configuration. Audio quality on the line is perfect during voice calls. No static or other noise. I've asked for disconnect supervision to be added to the line, but It doesn't look like it's there. The line still seems to keep the channel open long after the far end hangs up. Has anyone else ever seen this problem or have any ideas how to eliminate it? Brent, I had a similar problem and I feel for you, its frustrating. Are you using polycom phones by chance? Here is the problem that I had, not sure if your problem is related. Specs: - 6 Polycom 301 phones. - CentOS 4 Server with Asterisk 1.2.x - Sangoma A200 card with 3 FXO ports. Pretty simple settings for a small office where a group ring between all 6 polycoms was initiated once the call was received. After that it would go to a auto attendant and give the caller option to continue to hold, leave a message, etc. At any rate, once in a while, Caller ID would fail, either on the Sangoma card or somewhere in Asterisk, not sure, but CLI clearly showed an error on caller ID every once in a while when a POTS call came in. All six polycoms would ring, but when you picked up the handset or hit the Answer soft button, nothing would happen, you couldn't answer the call. The phones would just ring, and ring and ring for the duration of the group ring (about 60) and the customer was really annoyed since it was a small office. Continuing, the problem finally turned out to be the polycoms! When no caller ID information was present, the polycoms wigged out and while they did ring, you could not get the phones to pick up. I could readily replicate the behavior by initiating a Call File without specifying the caller ID information using the local channel. It would happen every time. Specifying the CID would allow the polycoms to work correctly. On the customer side, I did a quick GoToIf in their dialplan to see if the caller id info was set and if it wasn't I would set it manually to something like: CALLERID(num)=555-555- CALLERID(name)=CID FAILURE That cleared up the problem. HIH -- Warm Regards, Lee ___ -- 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] Phantom Rings
Lee Jenkins wrote: Brent, I had a similar problem and I feel for you, its frustrating. Are you using polycom phones by chance? Here is the problem that I had, not sure if your problem is related. Specs: - 6 Polycom 301 phones. - CentOS 4 Server with Asterisk 1.2.x - Sangoma A200 card with 3 FXO ports. Pretty simple settings for a small office where a group ring between all 6 polycoms was initiated once the call was received. After that it would go to a auto attendant and give the caller option to continue to hold, leave a message, etc. At any rate, once in a while, Caller ID would fail, either on the Sangoma card or somewhere in Asterisk, not sure, but CLI clearly showed an error on caller ID every once in a while when a POTS call came in. All six polycoms would ring, but when you picked up the handset or hit the Answer soft button, nothing would happen, you couldn't answer the call. The phones would just ring, and ring and ring for the duration of the group ring (about 60) and the customer was really annoyed since it was a small office. Continuing, the problem finally turned out to be the polycoms! When no caller ID information was present, the polycoms wigged out and while they did ring, you could not get the phones to pick up. I could readily replicate the behavior by initiating a Call File without specifying the caller ID information using the local channel. It would happen every time. Specifying the CID would allow the polycoms to work correctly. On the customer side, I did a quick GoToIf in their dialplan to see if the caller id info was set and if it wasn't I would set it manually to something like: CALLERID(num)=555-555- CALLERID(name)=CID FAILURE That cleared up the problem. HIH -- Warm Regards, Lee That really doesn't surprise me with Polycoms. They have excellent voice quality but their interface is dismal. I'm using Snom phones and I'm fairly certain the problem is with the phone line. I have all callerID settings disabled as the Telco is unable to provide it along with our rollover line setup due to limitations in their antiquated switch. The CLI and Logs all plainly show the calls as if they were normal calls with the exception of a message about Failed to write frame and no DTMF attempts, then the call is routed into the operator queue. The calls always came in on Zap1-1 so I tried swapping the 2 lines to see if it stayed on port 1 or if the phantom followed the line. As expected, the phantom rings followed the line and began showing up on Zap2-1. So it pretty has to be something in the telco, but I'm not sure what. Putting WaitForRing(3) before the Answer command in my IVR menu eliminates most of them, but sometimes more of them slip through. -Brent ___ -- 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
