Re: [Asterisk-Users] Freak incidents, who's to blame?
On 3-May-05, at 9:16 PM, Henry Devito wrote: Nortel and Toshiba and so on help eliminate this by routing outgoing calls starting from the highest trunk backwards and incoming calls of course start from the lowest trunk and work upward. Thanks for everyone's feedback on this. Just to add closure to the discussion, Zap channel groups support this hunting by use of a group number prefix in the dial cmd: http://voip-info.org/wiki-Asterisk+ZAP+channels Dial(Zap/g2...): Looks in order 1, 2, 5, 8 Dial(Zap/G2...): Looks in order 8, 5, 2, 1 Dial(Zap/r2...): Looks in order 8, 1, 2, 5 Dial(Zap/R2...): Looks in order 2, 1, 8, 5 (I was previously only aware of g). Thanks Ryan Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? It's called glare. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
On May 5, 2005 10:05 am, Ryan Courtnage wrote: Dial(Zap/g2...): Looks in order 1, 2, 5, 8 Dial(Zap/G2...): Looks in order 8, 5, 2, 1 Dial(Zap/r2...): Looks in order 8, 1, 2, 5 Dial(Zap/R2...): Looks in order 2, 1, 8, 5 Let's just be clear. Round Robin (r and R) will go through all the channels in order (ascend or decend) before reusing a channel. group (g and G) will find the first available channel (lowest or highest). So if you are using r, the order is always 1 2 3 4 5 6... and will not use 1 again until all of the others have been used. And if you are using g, the order will be the same BUT if a lower-numbered channel becomes free again it will be selected instead of the next numeric ordered channel. e.g. call out: uses 1 call out: uses 2 call out: uses 3 (call on 1 finishes) call out: uses 1 call out: uses 4 call out: uses 5 (call on 4 finishes) call out: uses 4 call out: uses 6 as you can see, it always uses the first available channel. G is the same but uses the highest numbered available channel. If you're trying to avoid glare, use g. If you're going for even usage then use r. And please, for the sake of the list, trim your responses; there is no need to include the ENTIRE email you're replying to. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
On Tue, 3 May 2005, Andrew Kohlsmith wrote: On May 3, 2005 02:22 pm, Ryan Courtnage wrote: From what I've read, glare is common in 2-way loopstart (kewlstart) circuits, and is impossible(?) to eliminate completely. But now I'm wondering what Nortel would tell a customer who experiences glare on their new Meridian system... they must do something to prevent glare from happening. Any ideas? Nope. Technically it shouldn't be possible with PRI but it is and does happen. Typically you hunt up starting at the highest available channel, and the telco hunts down which tends to keep it at bay until things get busy. Glare is when both the net and the cpe end attempt to seize a line simultaneously and both believe they succeeded. Glare really is impossible on a pri as a B channel can not be requested and allocated to both parties by mistake. The handshaking performed leaves no ambiguity as to which call a line is allocated to. However, a similar situation can occur when the cpe end requests a specific B channel in a SETUP message instead of leaving the channel selection to the net end. Unlike the glare condition this situation is detected and the net end prevails. The cpe end should then try to allocate another B channel with a new SETUP message. Unfortunatly Asterisk as a cpe device neither lets the net end allocate the B channel, nor does it retry using a different B channel. The problem is that Asterisk does not see the whole PRI as a single link with several channels, it sees the inidvidual channels with a common signalling path. A specific B channel is allocated before the signalling starts. This is a deficiency in Asterisk, not in isdn in general. The solution for Asterisk is the same as for glare-prone links - hunt for channels in the opposite direction. Note that on isdn links quite a few operators will by default _not_ hunt from one end or another, this has to be requested. The convention then is for the net end to hunt low-to-high and the cpe end to hunt high-to-low. Finally, even on isdn you have end devices (phones) which may themselves be prone to the human equivalent of glare - picking up the handset before the ring is heared. Some phones allow the user to request an outside line by pressing a button to prevent this. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
On May 4, 2005 02:54 am, Peter Svensson wrote: Unfortunatly Asterisk as a cpe device neither lets the net end allocate the B channel, nor does it retry using a different B channel. The problem is that Asterisk does not see the whole PRI as a single link with several channels, it sees the inidvidual channels with a common signalling path. A specific B channel is allocated before the signalling starts. This is a deficiency in Asterisk, not in isdn in general. Wow, thank you for this very insightful response... It's concise and describes exactly what the problem is, and why. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? I dont know who to blame, but we've had the same problem here with our small sales team. The sales team (about once a week) will dial a call on their analog phones (analog cordless phones plugged into a few SPA-2001s) - they press 'talk', dial the #, then immediatly are connected to an incomming call... (I use two TDM quad FXO cards to service 8 incomming lines from Sprint). I havnt been able to track it down, and its not reproducable manually... From the description, it almost sounds like glare. With analog fxo lines, that essentially means that both asterisk and the telco central office attempted to use the same pstn line for outgoing and incoming lines at the same time. Statistically, glare will occur more frequently with _small_ numbers of pstn lines and _greater_ amounts of traffic. I'd also guess that part of the problem might relate to how asterisk handles call setup. In other words, when an incoming call arrives at asterisk, asterisk probably doesn't mark the line as busy until after the callerid arrives (and the first internal ring occurs). If an out- going call is initiated at that time, asterisk may not know an incoming call is just arriving. But, that's a guess for sure. Might try using immediate=yes and usecallerid=no to see if that has any impact. If it does, then suspect the above timing issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. I don't think this is a freak incident at all. It still happens to me with people I call frequently and is easily explainable. you make a call, the telco connects it, and before the ring generator comes into a phase of putting voltage on the line, they pick up the phone. The circuit was connected, it just never got a chance to ring, there is nothing freak about it, just a matter of timing. Might also add that most central office switches do not sync the ringback audio with the actual ringing of the pstn line. So, ringback in many cases may be several seconds before/after the actual pstn line is ringing. Listening for ringback will not be a valid indicator of anything just in case someone suggests doing that. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Freak incidents, who's to blame?
Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? Thanks Ryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? It's called glare. http://home.intekom.com/scotland/cookbook/146.htm http://www.authorizedcom.com/lines_trunks.asp http://www.beagle-ears.com/lars/engineer/telecom/bizphone.htm http://www.zvon.org/tmRFC/RFC3064/Output/chapter4.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
On Tuesday 03 May 2005 11:40 am, Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? I dont know who to blame, but we've had the same problem here with our small sales team. The sales team (about once a week) will dial a call on their analog phones (analog cordless phones plugged into a few SPA-2001s) - they press 'talk', dial the #, then immediatly are connected to an incomming call... (I use two TDM quad FXO cards to service 8 incomming lines from Sprint). I havnt been able to track it down, and its not reproducable manually... Anybody have any ideas? -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. I don't think this is a freak incident at all. It still happens to me with people I call frequently and is easily explainable. you make a call, the telco connects it, and before the ring generator comes into a phase of putting voltage on the line, they pick up the phone. The circuit was connected, it just never got a chance to ring, there is nothing freak about it, just a matter of timing. This same thing could apply to asterisk or any other pbx or telco system since they all use the same basic arrangement of connecting circuits and ringing them. There is no hard and fast rule I know of that says a line must ring at least once before the called station can go off hook. Modems and such can be programmed with this rule, but a person picking up can do so at any random time. An exception would be when there is more than one call path to a phone, and effectively picking up to dial is not the same as picking up to answer an incoming call. Also, this could start happening more, not less as technology progresses since something like camp-on would have higher odds of hitting this situation since it constantly retries the number instead of the randomness at both ends if humans are involved. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? Thanks Ryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4277a74f32083462913418! 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
On May 3, 2005 11:40 am, Ryan Courtnage wrote: Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. It's not a freak accident; the switch just routed a call to you the exact moment that you picked up the line. It happens. My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? Avoidable? Not really; it happens. Is it really that much of a problem? Can I ask why? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
-Original Message-
From: Ryan Courtnage [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 03, 2005 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Freak incidents, who's to blame?
Hello all,
Everyone has probably experienced this at some point in the past:
You pick up your analog phone. Rather than hearing dialtone,
you are connected with someone who has just called you. Neither you
nor them heard a ring.
{clip}
My questions are: Who's to blame (telco, tdm card, * config,
gremlins)? Is this avoidable?
This is a condition known as 'glare' and is expected behaviour and is actually
quite common on busy lines unless a handshaking protocol of some sort is use
(such as wink or ground start). Essentially this handshake signals an intent to
use the trunk before it's actually seized, thereby preventing both parties from
seizing it simultaneously.
Google can provide far more detail.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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Re: [Asterisk-Users] Freak incidents, who's to blame?
Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? Thanks Ryan Really no one is to blame This is known as Glare, or a head on ( collision ) Take a basic Telephony course before attempting to become a telecom engineer. Back in the good old days a PBX would have analog trunks that were ground start, and tip was open when idle. The PBX would have an interface that knew this, and if it found ground on the tip would move to the next outgoing trunk. Ground was applied to the trunk by the CO BEFORE it was rung. The ring signal would then start whatever sequence in the PBX was necessary to answer the call. I gather that the FXO card doesn't support a ground start trunk at all, being nothing more than a modem given a new lease on life. Something other than a X100 or TDM400 is needed, and analog Ground start trunks from the Telco are called for for proper two way operation. GS trunks are certainly still available in the US and probably elsewhere. If you need enough trunks, perhaps a PRI instead? John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
On 3-May-05, at 10:34 AM, Eric Wieling aka ManxPower wrote: Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? It's called glare. Thank you, I'm now walking down the right path. From what I've read, glare is common in 2-way loopstart (kewlstart) circuits, and is impossible(?) to eliminate completely. But now I'm wondering what Nortel would tell a customer who experiences glare on their new Meridian system... they must do something to prevent glare from happening. Any ideas? Ryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Tuesday, May 03, 2005 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Freak incidents, who's to blame? Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? It's called glare. All elaborate on 'GLARE' Think about opening your front door on your way out just before the person gets a chance to ring the doorbell. Bam, there you are just GLARING at on e another. Glare could probably be reduced if you ALWAYS walked out the back door and those coming to visit you go to the front. You can set * to select lines in an order opesite than the way they roll over. For example our lines ring in ascending order line 1, then 2, 3,4,5, etc. If you set the PBX to grab line in decending order you will make outgoing calls on lines 5,4,3,2,1, decreasing the chances that * will open a ZAP door and find someone 'standing' there Hope this helps. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
If you're going through a CLEC for your lines, they can probably set the Glare Preference to be You or the Telco. I'm not sure if the Baby Bells would add that preference option for you. -m On Tue, 3 May 2005, Eric Wieling aka ManxPower wrote: Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? It's called glare. http://home.intekom.com/scotland/cookbook/146.htm http://www.authorizedcom.com/lines_trunks.asp http://www.beagle-ears.com/lars/engineer/telecom/bizphone.htm http://www.zvon.org/tmRFC/RFC3064/Output/chapter4.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
Really no one is to blame This is known as Glare, or a head on ( collision ) Take a basic Telephony course before attempting to become a telecom engineer. Back in the good old days a PBX would have analog trunks that were ground start, and tip was open when idle. The PBX would have an interface that knew this, and if it found ground on the tip would move to the next outgoing trunk. Ground was applied to the trunk by the CO BEFORE it was rung. The ring signal would then start whatever sequence in the PBX was necessary to answer the call. I gather that the FXO card doesn't support a ground start trunk at all, being nothing more than a modem given a new lease on life. Something other than a X100 or TDM400 is needed, and analog Ground start trunks from the Telco are called for for proper two way operation. GS trunks are certainly still available in the US and probably elsewhere. If you need enough trunks, perhaps a PRI instead? I don't think it is even that simple to get rid of. GS trunks would take care of the remote connect issues, but the same issue is still going to exist on the loopstart loops to the phones themselves. if you dial a local extension the other person can still pickup before the first ring if you are using analog phones. All this does is move the problem with analog trunks into a local only situation, but it still exists. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4277dc0474196392812593! 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
Ryan Courtnage wrote: On 3-May-05, at 10:34 AM, Eric Wieling aka ManxPower wrote: Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? It's called glare. Thank you, I'm now walking down the right path. From what I've read, glare is common in 2-way loopstart (kewlstart) circuits, and is impossible(?) to eliminate completely. But now I'm wondering what Nortel would tell a customer who experiences glare on their new Meridian system... they must do something to prevent glare from happening. Any ideas? They would tell them to use groundstart or EM wink ports, and then tell them that the loopstart cards they bought won't work with the other signaling methods and have to be replaced. --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
On May 3, 2005 02:22 pm, Ryan Courtnage wrote: From what I've read, glare is common in 2-way loopstart (kewlstart) circuits, and is impossible(?) to eliminate completely. But now I'm wondering what Nortel would tell a customer who experiences glare on their new Meridian system... they must do something to prevent glare from happening. Any ideas? Nope. Technically it shouldn't be possible with PRI but it is and does happen. Typically you hunt up starting at the highest available channel, and the telco hunts down which tends to keep it at bay until things get busy. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
Ryan Courtnage wrote: On 3-May-05, at 10:34 AM, Eric Wieling aka ManxPower wrote: It's called glare. Thank you, I'm now walking down the right path. From what I've read, glare is common in 2-way loopstart (kewlstart) circuits, and is impossible(?) to eliminate completely. But now I'm wondering what Nortel would tell a customer who experiences glare on their new Meridian system... they must do something to prevent glare from happening. Any ideas? Yup. Order your lines groundstart. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Freak incidents, who's to blame?
Nortel and Toshiba and so on help eliminate this by routing outgoing calls starting from the highest trunk backwards and incoming calls of course start from the lowest trunk and work upward. - Original Message - From: Ryan Courtnage [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 1:22 PM Subject: Re: [Asterisk-Users] Freak incidents, who's to blame? On 3-May-05, at 10:34 AM, Eric Wieling aka ManxPower wrote: Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? It's called glare. Thank you, I'm now walking down the right path. From what I've read, glare is common in 2-way loopstart (kewlstart) circuits, and is impossible(?) to eliminate completely. But now I'm wondering what Nortel would tell a customer who experiences glare on their new Meridian system... they must do something to prevent glare from happening. Any ideas? Ryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
