Re: [Asterisk-Users] Anyone using * in a live production environment?
Gavin Hamill wrote: Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case study, just a few lines to try and get a grip on the number of real installations. Thank you :) Gavin. Hi Gavin, If its possible could I get a copy of the business case?? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 10:04, WipeOut wrote: > Gavin Hamill wrote: > >I'm simply trying to add weight to the business case for new * installs, > >especially for those who have a very conservative management structure. > Hi Gavin, > > If its possible could I get a copy of the business case?? Ah, I can see how that sentence was miseleading... I'm talking rhetorically about the general "business case" for using * at all, as opposed to an actual document that I guess you thought I was referring to. If I already had something documented, it would certainly be publically available, since it benefits us all. Sorry :) gdh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone using * in a live production environment?
Me too! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Tuesday, November 04, 2003 1:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Anyone using * in a live production environment? Gavin Hamill wrote: >Hullo again, all :) > >If you're using * to run telephony in a real business environment, can I >trouble you to write a short paragraph about the setup, and how you've >found the migration / daily use? > >I'm simply trying to add weight to the business case for new * installs, >especially for those who have a very conservative management structure. > >Like I say, I'm not looking for a case study, just a few lines to try >and get a grip on the number of real installations. > >Thank you :) > >Gavin. > > > Hi Gavin, If its possible could I get a copy of the business case?? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Gavin Hamill wrote: On Tue, 2003-11-04 at 10:04, WipeOut wrote: Gavin Hamill wrote: I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Hi Gavin, If its possible could I get a copy of the business case?? Ah, I can see how that sentence was miseleading... I'm talking rhetorically about the general "business case" for using * at all, as opposed to an actual document that I guess you thought I was referring to. If I already had something documented, it would certainly be publically available, since it benefits us all. Sorry :) gdh No problem, Maybe I should have read it closer.. Well then if you do put one together I would be interested in seeing that one.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
-- Original Message -- From: Gavin Hamill <[EMAIL PROTECTED]> >Hullo again, all :) > >If you're using * to run telephony in a real business environment, can I >trouble you to write a short paragraph about the setup, and how you've >found the migration / daily use? > >Thank you :) > >Gavin. > Ok here is a short paragraph on our use of Asterisk in the real world. 1 inbound PRI ISDN 23/24 channel - Local phone service with 60 DID numbers. 1 Long Distance T1 line for inbound 800 numbers and all outbound long distance calls. running on P4 1.7G 512mg 20 gig HDD with 2 T400P boards. Ethernet on MB Intel 845G board. Hardware: 4 Adtran 750 with 24 FXS channels each. 1 Adtran 600 with 4 FX0 and 12 FXS ports. 1 ZetaFax server with US robotics modem. 1 HP Fax as backup 4 Inbound RAS lines for users 2 outbound RAS modems for dial out support lines. 40 452 phones (Really bad choice for phones) 10 390 phones (Again better then 452 but still bad phones) Cisco ATA 186 (nice works great) Cisco 7960 (Nice phone but worst phone to setup and maintain) 4 SIP phones X-Ten Lite with Telex USB connection to PC (Works great) Overall system is working with Support queues(AGI login user accounts) and meeting rooms. Voicemail system is not very good need some way to configure the boxes. They really need to redo this application for more standard settings. We have MOH working without any problems. Major down is no Graphical interface. No actual working manager. Got to get them to fix the Zombie lines. (I feel it's mainly do to our 390 and 452 phones. It works needs some fine tuning but it works. I have nothing good to say about the Aastra phones 390 or the 452. They are not really good for heavy use like we need! The Cisco 7960 is nice to look at but in the real world it's hard to get working and setup. If you don't know about Linux or are able to use scripts it's a real mess to keep up! This is where it's being held back as a real world player! This is the basic setup. Next step is outside offices connection. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 03:48, Gavin Hamill wrote: > Hullo again, all :) > > If you're using * to run telephony in a real business environment, can I > trouble you to write a short paragraph about the setup, and how you've > found the migration / daily use? > > I'm simply trying to add weight to the business case for new * installs, > especially for those who have a very conservative management structure. > > Like I say, I'm not looking for a case study, just a few lines to try > and get a grip on the number of real installations. I'm not trying to flame you for this message, but this is something that is asked about quite often and doesn't exactly prove anything. Just because I successfully pulled off an installation doesn't mean you will be successful. There is obviously working systems out here otherwise there wouldn't be this much traffic on this list. Maybe what needs to happen to keep this question from coming up over and over again would be to get this on the wiki. Someone want to create the page and then post the link to that specific page to be filled in? -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Ariel Batista wrote: Ok here is a short paragraph on our use of Asterisk in the real world. 1 inbound PRI ISDN 23/24 channel - Local phone service with 60 DID numbers. 1 Long Distance T1 line for inbound 800 numbers and all outbound long distance calls. running on P4 1.7G 512mg 20 gig HDD with 2 T400P boards. Ethernet on MB Intel 845G board. Hardware: 4 Adtran 750 with 24 FXS channels each. 1 Adtran 600 with 4 FX0 and 12 FXS ports. 1 ZetaFax server with US robotics modem. 1 HP Fax as backup 4 Inbound RAS lines for users 2 outbound RAS modems for dial out support lines. 40 452 phones (Really bad choice for phones) 10 390 phones (Again better then 452 but still bad phones) Cisco ATA 186 (nice works great) Cisco 7960 (Nice phone but worst phone to setup and maintain) 4 SIP phones X-Ten Lite with Telex USB connection to PC (Works great) Overall system is working with Support queues(AGI login user accounts) and meeting rooms. Voicemail system is not very good need some way to configure the boxes. They really need to redo this application for more standard settings. We have MOH working without any problems. Major down is no Graphical interface. No actual working manager. Got to get them to fix the Zombie lines. (I feel it's mainly do to our 390 and 452 phones. It works needs some fine tuning but it works. I have nothing good to say about the Aastra phones 390 or the 452. They are not really good for heavy use like we need! The Cisco 7960 is nice to look at but in the real world it's hard to get working and setup. If you don't know about Linux or are able to use scripts it's a real mess to keep up! This is where it's being held back as a real world player! This is the basic setup. Next step is outside offices connection. ___ What kind of resources are being consumed on the server? CPU,MEM,DISK,etc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 14:56, Ariel Batista wrote: > Ok here is a short paragraph on our use of Asterisk in the real world. Thank you Ariel - this is exactly the sort of information I am looking for :) I'm assuming this is actually for Avionica rather than you just writing from a different e-mail address? >From our own point of view, the lack of UI and dependency on scripting is no problem since we already have a Perl/PHP programming contingent here, so your words are actually encouraging :) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
-- Original Message -- From: Ken Godee <[EMAIL PROTECTED]> >Ariel Batista wrote: >> Ok here is a short paragraph on our use of Asterisk in the real world. >> > ___ > >What kind of resources are being consumed on the server? >CPU,MEM,DISK,etc I am not a Linux person (Trying to learn) so I am not able to check this out! But I do have over 12 gig of disk space still available. If you have some program or setting I can run on the server to give me this info I would love to see it! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 15:08, Steven Critchfield wrote: > I'm not trying to flame you for this message, but this is something that > is asked about quite often and doesn't exactly prove anything. Just > because I successfully pulled off an installation doesn't mean you will > be successful. No offence taken. Actually, I agree entirely. My intent is purely to show management that * is something worth allocating my time to research further, and not just some uber-geek project that's "fun, but of no commercial value." > Maybe what needs to happen to keep this question from coming up over and > over again would be to get this on the wiki. Someone want to create the > page and then post the link to that specific page to be filled in? Putting these on the Wiki isn't a bad idea, and is something I considered, but given the number of problems that people seem to have with it (e.g. it works fine, but is dead-slow from here...), and that e-mail reply takes fewer brain-cycles than registering for an acct, and formatting a Wiki posting, etc. People are doing the community a favour of providing the information, so it's in everyone's interests to make it as painless a process as possible. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Steven Critchfield wrote: On Tue, 2003-11-04 at 03:48, Gavin Hamill wrote: Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case study, just a few lines to try and get a grip on the number of real installations. I'm not trying to flame you for this message, but this is something that is asked about quite often and doesn't exactly prove anything. Just because I successfully pulled off an installation doesn't mean you will be successful. There is obviously working systems out here otherwise there wouldn't be this much traffic on this list. Maybe what needs to happen to keep this question from coming up over and over again would be to get this on the wiki. Someone want to create the page and then post the link to that specific page to be filled in? Actually, this is the thing I find most usefull. There are not very many posts where the statement is about a fully functional asterisk system being deployed in a business. The vast majority of posts are about problems with asterisk, and in my case it has kept me from replacing our functional but aging PBX with asterisk. More success stories with details about the hardware and the environment would be of great benefit to this list, in my opinion. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Ariel Batista wrote: -- Original Message -- From: Ken Godee <[EMAIL PROTECTED]> Ariel Batista wrote: Ok here is a short paragraph on our use of Asterisk in the real world. ___ What kind of resources are being consumed on the server? CPU,MEM,DISK,etc I am not a Linux person (Trying to learn) so I am not able to check this out! But I do have over 12 gig of disk space still available. If you have some program or setting I can run on the server to give me this info I would love to see it! ___ Quick and dirty from console prompt you can use "top" From a desktop you can try ie.. "xosview" Or install something like "gkrellm" http://www.gkrellm.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Gavin Hamill wrote: Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? Well, it's not technically a business, but we are running "*" for our research lab with the following hardware: 4 analog POTS lines ADTRAN TSU-600 FXO and FXS Plugins 20 GS budgetone 100, 2 Analog phones Asus A7V, 900MHz Athalon, 256 MB Ram, 60GB hard disk 100BaseT Ethernet environment using BayStack 450 switches Digium T100P RH9 APC Smart-Ups 700 Notes: The choice of the channel bank was based on the possibility of future expansion for handling several other labs (and the fact that I got is and 6-V.35 cards, which I', not using) for $99 on e-bay). The installed infrastructure for the phone system at our University dictated our choice of POTS lines. We were also not given any choice about the Baystack switches . The Budgetone phones are probably not the ideal solution for a larger installation (yet), but met our needs perfectly, perform very well and are easy to set up and install. The analog phones are to provide emergency service during power failures. The UPS is sized small because we have emergency power failover after about 30 seconds. The CPU and MOBO choices were based on experience with the products. A word of warning: Echo WILL be a potential problem on any system that has a transition point from POTS 2-wire lines to a TDM environment (see my previous posting). Here are some suggestions about dealing with it: 1) Use the highest possible CPU speed. Why: The single greatest problem with software echo cancellation is the computational intensity of the autocorrelation needed to determine the amount of echo at each discrete delay time. If the CPU is slow, the canceller will not be able to keep up with the real time nature of the computations. 2) Enable MMX options (assuming that you have a CPU which supports it). Why: Same as in item 1. In my case, this provided the single greatest improvement. 3) Lower the number of taps in the echo canceller. Why: While this may seem counter intuitive, since the number of taps determines the amount of delay the canceller can deal with, most of the time the delay you hear is much longer than the delay seen by the echo canceller. Decreasing the number of taps reduces the size of the autocorrelation and reduces the comptational load. This allows the canceller to learn faster and keep up more easily. 4) Turn off the KDE, etc. Why: Saves computation cycles for the echo cancellation. This is no joke, you can get on the phone, start up the desktop and hear the echo return. 5) Attempt to balance the hybrid at the 2-line to 4 line interface. Why: 99% of the time, this is where the echo originates and this is where is should be fixed. Unfortunately, this is not for the faint of heart, but if your line card has a hybrid balance adjustment (many don't), use it. Also, with multiple simultaneous calls, this may be the only real solution. Part of the problem arises from the use of lower impedance telephone wiring nowdays. The typical characteristic impedance of Cat5 twisted pair is about 100 ohms and many line cards are optimized for a 600 ohm line. This is made worse if the DC resistance of the wiring to the CO switch is relatively low. I haven't tried this myself, but you might try something as simple as a 500 ohm variable resistor in series with the ring line and adjust for minimum echo. If it gets worse, you haven't lost anything, just take the resistor out of the line. If it works, measure the value of the resistor when set for minimum echo and replace it with a fixed value resistor. 6) Try messing with Tx and Rx gains. Why: This is a reincarnation of the technique used to cancel echo on long lines in the early part of the 20th century. The idea is that the perception of echo gets worse as echo volume increases and as delay increases. You can sometimes reduce echo to an "acceptable" level if you attenuate it enough, especially if the echo canceller takes care of part of it. The problem is that you will likely run into unacceptably low volume levels. That's why this technique was abandoned by the PSTN: You've all seen the movies of people in the 1920's yelling into phones on long distance calls! Finally, while I had to do some head scratching and a lot of reading to get the system set up, I would have to say that the installation went without any major problems. Once configured, it runs flawlessly and requires very little maintainence. Steve Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 4 Nov 2003, Stephen R. Besch wrote: > 5) Attempt to balance the hybrid at the 2-line to 4 line interface. > Why: 99% of the time, this is where the echo originates and > this is where is should be fixed. Unfortunately, this is not for the > faint of heart, but if your line card has a hybrid balance adjustment > (many don't), use it. Also, with multiple simultaneous calls, this may > be the only real solution. Part of the problem arises from the use of > lower impedance telephone wiring nowdays. The typical characteristic > impedance of Cat5 twisted pair is about 100 ohms and many line cards are > optimized for a 600 ohm line. This is made worse if the DC resistance of > the wiring to the CO switch is relatively low. I haven't tried this > myself, but you might try something as simple as a 500 ohm variable > resistor in series with the ring line and adjust for minimum echo. If > it gets worse, you haven't lost anything, just take the resistor out of > the line. If it works, measure the value of the resistor when set for > minimum echo and replace it with a fixed value resistor. I've had to do similar things to lower loop current. Be sure you get at least 1 watt resistors and put the same size on the tip and ring. Putting resistance on only one wire will throw other things off. dave -- Dave Weis "I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Putting these on the Wiki isn't a bad idea, and is something I considered, but given the number of problems that people seem to have with it (e.g. it works fine, but is dead-slow from here...), and that e-mail reply takes fewer brain-cycles than registering for an acct, and formatting a Wiki posting, etc. Keep feeding the list, I'll steel information to the wiki. http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+medium+office If you need help with the Wiki, I'm available. I hope that with the recent changes in DNS and in database connectivity, the Wiki will be more available too. :-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
-- Original Message -- From: Ken Godee <[EMAIL PROTECTED]> >Ariel Batista wrote: > >>> >>>What kind of resources are being consumed on the server? >>>CPU,MEM,DISK,etc >> >Quick and dirty from console prompt you can use "top" > From a desktop you can try ie.. "xosview" >Or install something like "gkrellm" >http://www.gkrellm.net Thank you just ran the Top. I have 5.4 % CPU, System is 4.2 % Memory use is 234045 with 254690 free! Swap file says 0 used. Which means I have too much power for it! P4 1.7Gig with 512M RAM and over 12gig free still on a 20 gig HDD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Stephen R. Besch wrote: Gavin Hamill wrote: Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? Well, it's not technically a business, but we are running "*" for our research lab with the following hardware: Great information - many thanks! http://www.voip-info.org/tiki-index.php?page=Asterisk%20setup%20research%20lab Question: What is an Adtrans TSU-400? A channel bank? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Question: What is an Adtrans TSU-400? A channel bank? The TSU-600 is a channel bank. It accepts up to 6 plug in cards (with quad cards, up to 24 lines) and bundles them onto a single T1 Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 12:24, Olle E. Johansson wrote: > Keep feeding the list, I'll steel information to the wiki. > http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+medium+office > > If you need help with the Wiki, I'm available. Okay, looked there, but my setup doesn't fit any of the prefilled in examples. Not to mention, since these are real working systems, it should also contain some key words about success stories. So I'll just post our config here. Machine name phone: SuperMicro 5011E 1200 PIII 256 megs memory 40 gig IDE HD T400P ADIT 600 Machine name pbx: Generic system 1100 Celeron 256 megs memory 80 gig IDE HD T100P Zhone Zplex 10b Description of deployment. - 1 PRI from Telcove(formerly Adelphia) into phone located at our Colocation provider. - 1 point to point T1 from Telcove for data traffic from Colocation to our office. - phone acts as a switch only, and has little local load. - phone splits off 1 DID to the local ADIT 600 for a legacy answering system. - phone sends the rest of our 19 DID lines to pbx via GSM compressed IAX2 protocol. - pbx does local extension work. - pbx is used as a development platform for our software to replace legacy answering system. - 23 incoming lines with 10-13 lines peak being used on the legacy system, and 2-4 lines peak being used as extensions. - Has been in nearly fault free operation for more than since 05-2002. --- The above system is in use at my office. What appears below is my home system that currently is not being used, but will soon be put back together. Machine name homepbx Generic system 1400+ Duron 256 megs memory 40 gig IDE HD T100P ADIT 600 - ADIT 600 is attached to the T100P and serves up mostly cordless phones in my home. - Current set up has 14 lines attached to a display board and 3 extensions wired to different rooms of the house. - Outside PSTN connectivity was achieved by attaching to the PSTN gateway at the office via GSM compressed IAX2 protocol over my cable modem. - Could make 2 and sometimes 3 calls simultaneously over cable modem. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Dave Weis wrote: On Tue, 4 Nov 2003, Stephen R. Besch wrote: 5) Attempt to balance the hybrid at the 2-line to 4 line interface. Why: 99% of the time, this is where the echo originates and this is where is should be fixed. Unfortunately, this is not for the faint of heart, but if your line card has a hybrid balance adjustment (many don't), use it. Also, with multiple simultaneous calls, this may be the only real solution. Part of the problem arises from the use of lower impedance telephone wiring nowdays. The typical characteristic impedance of Cat5 twisted pair is about 100 ohms and many line cards are optimized for a 600 ohm line. This is made worse if the DC resistance of the wiring to the CO switch is relatively low. I haven't tried this myself, but you might try something as simple as a 500 ohm variable resistor in series with the ring line and adjust for minimum echo. If it gets worse, you haven't lost anything, just take the resistor out of the line. If it works, measure the value of the resistor when set for minimum echo and replace it with a fixed value resistor. I've had to do similar things to lower loop current. Be sure you get at least 1 watt resistors and put the same size on the tip and ring. Putting resistance on only one wire will throw other things off. dave The 1 watt recommendation is a good one. The split resistor will probably not make any difference on a voice only line, as the transformed impedance of the phone line on the other side of the hybrid won't really care. On the other hand, a big R imbalance will affect transmission line characteristics on the line, which will have a negative impact on data transmission. The bottom line is that resistors are cheap and using two is probably worth the extra pennies. Just be sure to divide the resistor value that minimizes echo in half so that the total resistance insterted is the measured value. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
IIRC, proper functioning of the 2-wire to 4-wire hybrid depends on proper balance between the the individual wires of the pair. If you upset the balance you're going to get all kind of problems. Along the same lines, overdriving the hybrid is a big source of echo. The correct way to set the RX/TX gain values would be to get the number for the "milliwatt test signal" from a friendly telco tech. You can dial that number through * and then adjust the RX gain value so that the signal is loud, but not overdriving the ADC on the analog interface card. A good starting point on the TX gain would be whatever the RX gain ends up at. The proper way to set TX gain would be to use a digital version of the miliwatt tone (1004Hz at 0dBm IIRC) outgoing and adjust it for a proper level through to another line on the same telco switch or across a T1 span that you can observe with a test set. I don't recall the exact specs for what percentage of the full-scale value milliwatt tones should be at on the digital systems. Anyone have the reference? THX/BDH On Tue, 2003-11-04 at 15:19, Stephen R. Besch wrote: > The 1 watt recommendation is a good one. The split resistor will > probably not make any difference on a voice only line, as the > transformed impedance of the phone line on the other side of the hybrid > won't really care. On the other hand, a big R imbalance will affect > transmission line characteristics on the line, which will have a > negative impact on data transmission. The bottom line is that resistors > are cheap and using two is probably worth the extra pennies. Just be > sure to divide the resistor value that minimizes echo in half so that > the total resistance insterted is the measured value. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Thank you, Steven! http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+success+1 ADIT 600 What is that? Zhone Zplex 10b And that? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 15:29, Olle E. Johansson wrote: > Thank you, Steven! > > http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+success+1 > > > ADIT 600 > What is that? Channel bank from Carrier Access. > > > Zhone Zplex 10b > And that? Channel Bank from Zhone. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Brian D Heaton wrote: IIRC, proper functioning of the 2-wire to 4-wire hybrid depends on proper balance between the the individual wires of the pair. If you upset the balance you're going to get all kind of problems. I just finished modelling a standard 4-transformer hybrid coupled to a balanced RC transmission line. Cross talk was zero when the hybrid was balanced. Inserting a single resistor in series with tip or ring imbalanced the hybrid and cross talk appeared. This could be completely compensated with the proper RC on the opposite side of the hybrid, as predicted. It made absolutely no difference to the cancellation if the resistor was split. Since a balanced hybrid appears as a pure resistance (complex terms are 0) to the transmission line, placing a simple resistor in series with the hybrid (on either side) at the termination point will just look like 2 resistors in series and will properly terminate the line. There should be no effects at all from doing this other than the loss of some energy in the termination resistor, which can be made up for with a boost in Rx gain. Along the same lines, overdriving the hybrid is a big source of echo. That's because the cores saturate on transformer based hybrids. This is not as likely to occur with active hybrids built with op-amps (which are found in almost all modern line cards), although it is possible if the gains are high enough. However the distortion from the clipping would be far worse than the echo. The correct way to set the RX/TX gain values would be to get the number for the "milliwatt test signal" from a friendly telco tech. You can dial that number through * and then adjust the RX gain value so that the signal is loud, but not overdriving the ADC on the analog interface card. A good starting point on the TX gain would be whatever the RX gain ends up at. The proper way to set TX gain would be to use a digital version of the miliwatt tone (1004Hz at 0dBm IIRC) outgoing and adjust it for a proper level through to another line on the same telco switch or across a T1 span that you can observe with a test set. I don't recall the exact specs for what percentage of the full-scale value milliwatt tones should be at on the digital systems. Anyone have the reference? THX/BDH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Stephen, Very interesting. I know I've seen all manner of messiness in the past when folks have monkeyed with balanced pairs. I'll take your word on the modeling data. I've not gone that far in depth with it. You don't have the specs on gain adjustment handy do you? I've probably got it buried in an old pub somewhere, but I don't have anything in soft-copy. THX/BDH On Tue, 2003-11-04 at 18:34, Stephen R. Besch wrote: > I just finished modelling a standard 4-transformer hybrid coupled to a > balanced RC transmission line. Cross talk was zero when the hybrid was > balanced. Inserting a single resistor in series with tip or ring > imbalanced the hybrid and cross talk appeared. This could be completely > compensated with the proper RC on the opposite side of the hybrid, as > predicted. It made absolutely no difference to the cancellation if the > resistor was split. Since a balanced hybrid appears as a pure > resistance (complex terms are 0) to the transmission line, placing a > simple resistor in series with the hybrid (on either side) at the > termination point will just look like 2 resistors in series and will > properly terminate the line. There should be no effects at all from > doing this other than the loss of some energy in the termination > resistor, which can be made up for with a boost in Rx gain. > > That's because the cores saturate on transformer based hybrids. This is > not as likely to occur with active hybrids built with op-amps (which are > found in almost all modern line cards), although it is possible if the > gains are high enough. However the distortion from the clipping would > be far worse than the echo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
- Original Message - From: "Ariel Batista" <[EMAIL PROTECTED]> > The Cisco 7960 is nice to look at but in the real world it's hard to get working and setup. > If you don't know about Linux or are able to use scripts it's a real mess to keep up! > This is where it's being held back as a real world player! While I will concede that the Cisco 7960 can be initially difficult to setup, provisioning of additional phones is a breeze once you have the initial configuration and services in place (things like DHCP, TFTP, etc). I purchased a few more recently, and can have additional phones talking to Asterisk in a matter of a minute or two - this includes the process of converting to SIP. The additional features are nice too. I'm making use of CCXML. You can press Directories -> external and get an employee directory, a list of call queues (convenient for transferring calls straight into a queue) and more. By pressing services, you can get access to a company quick reference which has things like our phone numbers, details on hosting plans, service addons, etc. - handy when you're on a call and don't have the info up on your computer screen. Overall it's a very functional phone and I certainly don't regret the expense in purchasing them (something I was initially worried about). -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
I'm in agree with all explanations regarding the echo and 2/4 wires conversion. However I'm wondering if there are other parameters like CPU and/or Asterisk configuration involved in the problem with more weight than hybrid. Otherwise how do you explain the difference in the following scenario: 1.- Crystal clear voice: [phone1][pabx]-[fxo gateway]--SIP---[fxs gateway]--[phone2] 2.- A lot of echo: [gnophone or xten]---[ * ]---SIP-[fxs gateway]-[phone2] The first scenario has four 2/4 W conversion. The second one has only two (or one?). The * was running in differents CPU, PIII 500 Mhz, PIII 750 Mhz, 128 Mb to 512 Mb ram with not difference on echo. We have installed a Mitel 3100 with IP phones at 40 kms within a wireless network with not echo at all. Jorge Stephen R. Besch wrote: Brian D Heaton wrote: IIRC, proper functioning of the 2-wire to 4-wire hybrid depends on proper balance between the the individual wires of the pair. If you upset the balance you're going to get all kind of problems. I just finished modelling a standard 4-transformer hybrid coupled to a balanced RC transmission line. Cross talk was zero when the hybrid was balanced. Inserting a single resistor in series with tip or ring imbalanced the hybrid and cross talk appeared. This could be completely compensated with the proper RC on the opposite side of the hybrid, as predicted. It made absolutely no difference to the cancellation if the resistor was split. Since a balanced hybrid appears as a pure resistance (complex terms are 0) to the transmission line, placing a simple resistor in series with the hybrid (on either side) at the termination point will just look like 2 resistors in series and will properly terminate the line. There should be no effects at all from doing this other than the loss of some energy in the termination resistor, which can be made up for with a boost in Rx gain. Along the same lines, overdriving the hybrid is a big source of echo. That's because the cores saturate on transformer based hybrids. This is not as likely to occur with active hybrids built with op-amps (which are found in almost all modern line cards), although it is possible if the gains are high enough. However the distortion from the clipping would be far worse than the echo. The correct way to set the RX/TX gain values would be to get the number for the "milliwatt test signal" from a friendly telco tech. You can dial that number through * and then adjust the RX gain value so that the signal is loud, but not overdriving the ADC on the analog interface card. A good starting point on the TX gain would be whatever the RX gain ends up at. The proper way to set TX gain would be to use a digital version of the miliwatt tone (1004Hz at 0dBm IIRC) outgoing and adjust it for a proper level through to another line on the same telco switch or across a T1 span that you can observe with a test set. I don't recall the exact specs for what percentage of the full-scale value milliwatt tones should be at on the digital systems. Anyone have the reference? THX/BDH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, Nov 04, 2003 at 07:24:19PM +0100, Olle E. Johansson wrote: > > Keep feeding the list, I'll steel information to the wiki. > http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+medium+office Superb stuff, Olle :) If we can establish a 'standard format' (maybe an HTML form?) for configuration postings, perhaps people will be more likely to submit their data? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, Nov 04, 2003 at 01:52:46PM -0600, Steven Critchfield wrote: > - Has been in nearly fault free operation for more than since 05-2002. Great stuff, Steven! :) Can I enquire what was the cause of the downtime? Was it planned- maintenance, or an actual fault with the Asterisk software / Digium hardware? Roughly how long has the system been down in total since 'going live' ? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Wed, 2003-11-05 at 03:11, Gavin Hamill wrote: > On Tue, Nov 04, 2003 at 01:52:46PM -0600, Steven Critchfield wrote: > > > - Has been in nearly fault free operation for more than since 05-2002. > > Great stuff, Steven! :) > > Can I enquire what was the cause of the downtime? Was it planned- > maintenance, or an actual fault with the Asterisk software / Digium > hardware? One failure was a kernel lockup when compiling a new module to start testing ZapRas. I have made a couple of stupid mistakes that shut asterisk down. Outside of the above comments, my gateway machine has worked flawlessly, but this is also why it isn't given much to do. It is too important to have something make it fail. On my pbx machine in the office though, we occasionally have small failures. Most recently we had a segfault show up in a zapata handle event function, but couldn't track it down well enough to report upon it. This machine is specifically set up to be more of a test bed machine. We have only 4 people currently in our office, and 2 of us use the phones mainly for testing of our software. > Roughly how long has the system been down in total since 'going live' ? I'd say we haven't had more than 10 minutes downtime on our gateway machine, and thats mostly due to the kernel lockup that caused me to have to call my colo facility to do a hands on reset of the machine. We may have about that much time on our pbx, but this is also where we test our patches to asterisk, so it can't be called against asterisk. My execution of this setup hasn't been telco quality, but seems pretty on par with small office pbx systems. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Brian, I think some of the confusion comes from what end of the line we are looking at and the nature of the imbalance. While the resistor may fix the near end, it will probably cause some termination problems at the far end. Reflections mostly, which on a short run analog line shouldn't be much of a problem. I haven't looked into this in detail. Also, if you look at the structure of a balanced transmission line, it should be really important to not have any imbalance out on the distributed part of the line, such as caused by having one wire of the pair having a different resistance, or by having a resistance anywhere but at the line termination - say 2000 feet out. If I interpret things correctly, this would give a line which has two termination resistances at which there is a peak of power transfer to the load, and neither of them would appear purely resistive, giving phase shift errors which make balancing the hybrid difficult or impossible, and degrading data transmission capability. Re the gain specs, I don't have a reference to them, but I suspect that there is stuff to be found on Google. Brian D Heaton wrote: Stephen, Very interesting. I know I've seen all manner of messiness in the past when folks have monkeyed with balanced pairs. I'll take your word on the modeling data. I've not gone that far in depth with it. You don't have the specs on gain adjustment handy do you? I've probably got it buried in an old pub somewhere, but I don't have anything in soft-copy. THX/BDH On Tue, 2003-11-04 at 18:34, Stephen R. Besch wrote: I just finished modelling a standard 4-transformer hybrid coupled to a balanced RC transmission line. Cross talk was zero when the hybrid was balanced. Inserting a single resistor in series with tip or ring imbalanced the hybrid and cross talk appeared. This could be completely compensated with the proper RC on the opposite side of the hybrid, as predicted. It made absolutely no difference to the cancellation if the resistor was split. Since a balanced hybrid appears as a pure resistance (complex terms are 0) to the transmission line, placing a simple resistor in series with the hybrid (on either side) at the termination point will just look like 2 resistors in series and will properly terminate the line. There should be no effects at all from doing this other than the loss of some energy in the termination resistor, which can be made up for with a boost in Rx gain. That's because the cores saturate on transformer based hybrids. This is not as likely to occur with active hybrids built with op-amps (which are found in almost all modern line cards), although it is possible if the gains are high enough. However the distortion from the clipping would be far worse than the echo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Jorge Mendoza wrote: I'm in agree with all explanations regarding the echo and 2/4 wires conversion. However I'm wondering if there are other parameters like CPU and/or Asterisk configuration involved in the problem with more weight than hybrid. Otherwise how do you explain the difference in the following scenario: 1.- Crystal clear voice: [phone1][pabx]-[fxo gateway]--SIP---[fxs gateway]--[phone2] 2.- A lot of echo: [gnophone or xten]---[ * ]---SIP-[fxs gateway]-[phone2] First, I assume that the right hand side in both cases is the same. Then determine where the echo originates. I would guess that it is either coming from the fxs gateway and that there is no echo cancellation running on the fxs, while there is on the fxo, or you are using an open mic and speaker on the softphone. In the first case, when the fxo is in the circuit, the echo canceller sees the echo - it doesn't care where it comes from, the signal will stii autocorrelate because it contains a copy of itself - and it gets removed. In the other case, the echo does not get removed because the echo canceller is not running on that channel. Examine the channel status (zap show channel x) and the canceller is active when a call is in progress. In the second case, use a headset. Echo cancellers are notoriously poor at cancelling room echo very well without very compute intensive algorithms and long tail lengths (lots of taps). It usually reguires a DSP. If it's neither of these, then I am stumped. The first scenario has four 2/4 W conversion. Well, I only see two possible hybrids here, unless the PABX is running analog at the external ports and digital internally -nutty, but possible. The second one has only two (or one?). Probably one. The * was running in differents CPU, PIII 500 Mhz, PIII 750 Mhz, 128 Mb to 512 Mb ram with not difference on echo. We have installed a Mitel 3100 with IP phones at 40 kms within a wireless network with not echo at all. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
> 5) Attempt to balance the hybrid at the 2-line to 4 line interface. This is _precisely_ why my rollouts are all strongly recommending using a channel bank instead of the cheap X100P/TDM400P cards -- a lot of work has been put into the hybrid circuitry to dynamically adjust to the line impedance. I've had no serious issues with the X100P/TDM400P in small scale stuff but the echo cancel IMO should be done where it originates -- at the hybrid. Having said that, I do have "echocancel=32" in my zapata.conf for the T100P connected directly to an Adit600 FXS channel bank. I also have an old CAC AB1 with 12FXS and 12FXO ports I am going to deploy shortly to test things like far-end disconnect and other issues. > be the only real solution. Part of the problem arises from the use of > lower impedance telephone wiring nowdays. The typical characteristic > impedance of Cat5 twisted pair is about 100 ohms and many line cards are > optimized for a 600 ohm line. This is made worse if the DC resistance of > the wiring to the CO switch is relatively low. I haven't tried this This is a neat idea; something I have not thought of. However my ideal PSTN termination is digital (PRI) ... something to eliminate the hybrid altogether, at least on my end. :-) For deployments where I am simply providing VOIP to an existing phone system, I am recommending installing a T100P and a digital trunk for the existing KSU; again to eliminate the hybrid mess, or at least push it off to someone else's problem. :-) > 6) Try messing with Tx and Rx gains. Something I have noticed is that on the Adit600 FXS ports, I have had to set its RX attenuation to -7dB!! (TX to -3dB) If my math is correct, that means I am attenuating 85% of my incoming signal! Is this perhaps what you are referring to with the super-low impedance? Thank you for this super technical and informative post. This is what *-users needs... more tech and less running around in circles with the same issues over and over! Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Stephen R. Besch wrote: 5) Attempt to balance the hybrid at the 2-line to 4 line interface. Why: 99% of the time, this is where the echo originates and this is where is should be fixed. Unfortunately, this is not for the faint of heart, but if your line card has a hybrid balance adjustment (many don't), use it. Also, with multiple simultaneous calls, this may be the only real solution. Part of the problem arises from the use of lower impedance telephone wiring nowdays. The typical characteristic impedance of Cat5 twisted pair is about 100 ohms and many line cards are optimized for a 600 ohm line. This is made worse if the DC resistance of the wiring to the CO switch is relatively low. I haven't tried this myself, but you might try something as simple as a 500 ohm variable resistor in series with the ring line and adjust for minimum echo. If it gets worse, you haven't lost anything, just take the resistor out of the line. If it works, measure the value of the resistor when set for minimum echo and replace it with a fixed value resistor. Tweaking the hybrid is really a waste of time. Most don't permit tweaking for this reason. Any change to the circuit, like changing to another phone (perhaps even of the same model) generally defeats the effect of any tweaking on the short lines of most PBXs. A well designed hybrid is fairly relaxed about termination, though the return loss can vary a lot across the audio band. Most approvals specs only call for about 12dB of return loss, and you will seldom see more than 20dB - even with hand tweaking. Whatever you do with the hybrid, only proper echo cancellation will clean things up well enough for good VoIP (or cellphone calls, which suffer similar high latency). Actually twisted pair are generally below 150ohms. No line can have an impedance higher than 377ohms. 600ohm termiation is a fudge. On long lines the fudge doesn't work well, and loading coils are needed to refudge things. ADSL can't work with the loading coils in place, so they are being stripped out in many places. Such is life - messy! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Andrew Kohlsmith wrote: 5) Attempt to balance the hybrid at the 2-line to 4 line interface. This is _precisely_ why my rollouts are all strongly recommending using a channel bank instead of the cheap X100P/TDM400P cards -- a lot of work has been put into the hybrid circuitry to dynamically adjust to the line impedance. God, I wish I could be so sure of that. I've looked at the circuitry on some high end channel bank active hybrids. SPICE modeling predicts a maximum 26 dB attenuation of the returned echo, even with a balanced line. A simple circuitry change which adds a balance adjustment should permit cancellation to approach the parctical limit of a good quality op-amp (60-100 dB). I don't know what the Digium stuff looks like, so I can't comment on it. I've had no serious issues with the X100P/TDM400P in small scale stuff but the echo cancel IMO should be done where it originates -- at the hybrid. Having said that, I do have "echocancel=32" in my zapata.conf for the T100P connected directly to an Adit600 FXS channel bank. Did it help? be the only real solution. Part of the problem arises from the use of lower impedance telephone wiring nowdays. The typical characteristic impedance of Cat5 twisted pair is about 100 ohms and many line cards are optimized for a 600 ohm line. This is made worse if the DC resistance of the wiring to the CO switch is relatively low. I haven't tried this This is a neat idea; something I have not thought of. However my ideal PSTN termination is digital (PRI) ... something to eliminate the hybrid altogether, at least on my end. :-) For deployments where I am simply providing VOIP to an existing phone system, I am recommending installing a T100P and a digital trunk for the existing KSU; again to eliminate the hybrid mess, or at least push it off to someone else's problem. :-) I hope someone tries it and reports back. I may do it myself if I can get the time. Hard to do when the system is up and running. 6) Try messing with Tx and Rx gains. Something I have noticed is that on the Adit600 FXS ports, I have had to set its RX attenuation to -7dB!! (TX to -3dB) If my math is correct, that means I am attenuating 85% of my incoming signal! Is this perhaps what you are referring to with the super-low impedance? I hate decibels! Technically, dB is defined either for power gain (10log(P/Pr)) or Voltage gain (20log(V/Vr)). They are related via the proportionality between power and voltage (P prop to V squared). When you move the square out of the log, it introduces the factor of 2 difference. The problem with all this is that many people are very sloppy about specifying whether they mean power gain or voltage gain. It is especially problematical when one is talking about line impedance matching, which affects power transfer. However, once you are past the hybrid and into the amplifiers, you are talkin about voltage gain and should use the '20log' formula. In this case, -7dB = 10^^(-7/20)=44.7%. In otherwords, you are losing 55.3% of the signal, just over half. The low impedance is another, related issue. Theoretically, when everyone is operating at the characteristic impedance of the line (which is determined by the capacitance of the wiring, spacing of the wires, wire gauge, insulation material, degree if twist, etc), all of the power delivered to the line at the sending end will be transferred to the line, and all of the power at the receiving end will be accepted by the load. Under any other conditions, some of the power will be reflected back onto the line. Delay is only a few ns per foot, so this doesn't amount to much in an analog only system with wires that are only a few miles long. In the present case, the real echo problem arises because of how the hybrid sees the line impedance. In order for the hybrid to work, it must subtract an exact copy of the transmitted signal from the received signal. Since there is inductance and capacitance involved, there are also phase shifts, so it isn't enough to just have a signal of the same amplitude. The phase also has to be matched. Mismatching the impedance throws off both the phase and the amplitude of the copy of the transmitted signal that is subtracted, resulting in imperfect cancellation and an echo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Stephen R. Besch wrote: God, I wish I could be so sure of that. I've looked at the circuitry on some high end channel bank active hybrids. SPICE modeling predicts a maximum 26 dB attenuation of the returned echo, even with a balanced line. A simple circuitry change which adds a balance adjustment should permit cancellation to approach the parctical limit of a good quality op-amp (60-100 dB). I don't know what the Digium stuff looks like, so I can't comment on it. The spice modeling is rather idealised if you get those figures. Few hybrids ever give more than 20dB. If they achieve more than 12dB across most of the audio band they pass the approvals. Most *only just* pass the approvals. What would be the point in getting great results at the channel bank, when the phone's own hybrid still gives you a -12dB echo? The phone tends to be worse than the other end. This is why VoIP systems have to use echo cancellation. Hybrids are for little more than keeping the feedback below the oscillation point. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
