From ratguy at insightbb.com Sat Dec 1 05:25:30 2007 From: ratguy at insightbb.com (Jayson Smith) Date: Sat, 1 Dec 2007 06:25:30 -0500 Subject: [VoIP] New Asterisk Speaking Clock AGI References: <20060331102639.2F9995613C@ns1.vyger.net> Message-ID: <000801c8340c$e1ae7d60$6900a8c0@BOE> Hi, This topic has been dead for almost two years! I just called 263-2525 yesterday afternoon, and see that apparently you haven't done a lot with it recently. Are you still planning at some point on making this code available? I've always wanted a truly accurate TOD service for Asterisk. I'd be interested in TIM-2000, except: A. it's apparently no longer available, B. it needs UK 50HZ AC power to keep time, and C. it sets itself with MSF, a UK time signal, instead of WWVB, the US equivalent. Has anyone ever thought of building a similar device for use in the US? I do have John Doyle samples available. Sadly, I know of no complete Jane Barbe speech sets that are available. On a somewhat related subject, I've always thought the Allison numbers and other similar words were spoken a bit slow for my liking. Are there good quality samples in existence where she speaks the numbers faster? I mean, where she *really* spoke them faster, not where someone's taken the normal samples and compressed them with a sound editor. Jayson ----- Original Message ----- From: "John R. Covert" To: "CNET" Sent: Friday, March 31, 2006 5:18 AM Subject: [VoIP] New Asterisk Speaking Clock AGI >I just foolishly stayed up all night and wrote a new Asterisk AGI > which does an accurate "at the third stroke" speaking clock. It's > synchronized by NTP, but suffers a slight VoIP delay. > > It's on CNET 1 263-2525. > > Sound file situation isn't all sorted out. I make the "At the > third stroke it will be" announcement myself, and then use > Allison for the rest. She speaks more slowly than I would like, > and this might actually be a problem from 2100-2359 for more than > a single cycle through the announcement. We'll see. I might need > to make a new full set of sound files which speak a bit faster. > > Or maybe the real BT sound files in ulaw or alaw might be available. > > /john > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > From voiptandem at shaneyoung.com Sun Dec 2 13:06:48 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Sun, 02 Dec 2007 13:06:48 -0600 Subject: [VoIP] OT: Looking for Parts Message-ID: <20071202130648.96lqabvqwwkcksk4@secure.shaneyoung.com> I've been hunting high and low for what seems to be a fairly common enclosure but can't seem to find the source. Here are a few different companies selling a product that looks like it has the same enclosure: http://tinyurl.com/2jhc9q http://tinyurl.com/3y3bpa http://tinyurl.com/2nhxhq This is a similar product: http://tinyurl.com/38dhzq I'm trying to find a source for just the enclosure with no electronics. Any hints or clues would be appreciated. --Shane +1-821-7311 CNET ---------------------------------------------------------------- From jjones3601 at yahoo.com Sun Dec 2 18:12:03 2007 From: jjones3601 at yahoo.com (john jones) Date: Sun, 2 Dec 2007 16:12:03 -0800 (PST) Subject: [VoIP] 72 pin EDO memory question Message-ID: <259987.69611.qm@web34314.mail.mud.yahoo.com> I'm trying to figure out why some 64M 72 pin EDO memory modules don't work in the Cisco MC3810. Cisco has always made a lot of memory by having internal part numbers with no cross reference numbers. The 3810 should take 60ns, 16X4 4k EDO, non-parrity modules according to one Cisco document I found. I thought this meant 16M x 4 bytes (32 bits) and the 4K refers to the refresh rate. The 64M modules I have all have 8 chips which seems to confirm the no-parity. Does anyone know what other parameters might exist? The 3810 uses the Motorola MPC860 processor which can use 8,16,or 32 bits. I don't know the router uses. Thanks! John From jnovack at stromberg-carlson.org Sun Dec 2 21:06:55 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Sun, 02 Dec 2007 22:06:55 -0500 Subject: [VoIP] 72 pin EDO memory question In-Reply-To: <259987.69611.qm@web34314.mail.mud.yahoo.com> References: <259987.69611.qm@web34314.mail.mud.yahoo.com> Message-ID: <475372CF.7010208@stromberg-carlson.org> The only thing I can add to the confusion is the 3810-V's I have are equipped with Kingston memory, KTM16X32LA-60EG 8 chips made by NEC and six little chips on the back side A part number label 9905051-003.A00 Not sure if that is Cisco or Kingston John Novack john jones wrote: > I'm trying to figure out why some 64M 72 pin EDO memory modules don't work in the Cisco MC3810. Cisco has always made a lot of memory by having internal part numbers with no cross reference numbers. The 3810 should take 60ns, 16X4 4k EDO, non-parrity modules according to one Cisco document I found. I thought this meant 16M x 4 bytes (32 bits) and the 4K refers to the refresh rate. The 64M modules I have all have 8 chips which seems to confirm the no-parity. > > > Does anyone know what other parameters might exist? The 3810 uses the Motorola MPC860 processor which can use 8,16,or 32 bits. I don't know the router uses. > > Thanks! > > John > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > -- Dog is my co-pilot From martin at Princeton.EDU Sun Dec 2 21:15:00 2007 From: martin at Princeton.EDU (Martin Harriss) Date: Sun, 02 Dec 2007 22:15:00 -0500 Subject: [VoIP] 72 pin EDO memory question In-Reply-To: <259987.69611.qm@web34314.mail.mud.yahoo.com> References: <259987.69611.qm@web34314.mail.mud.yahoo.com> Message-ID: <475374B4.4060000@Princeton.EDU> john jones wrote: > I'm trying to figure out why some 64M 72 pin EDO memory modules don't work in the Cisco MC3810. Cisco has always made a lot of memory by having internal part numbers with no cross reference numbers. The 3810 should take 60ns, 16X4 4k EDO, non-parrity modules according to one Cisco document I found. I thought this meant 16M x 4 bytes (32 bits) and the 4K refers to the refresh rate. The 64M modules I have all have 8 chips which seems to confirm the no-parity. > > > Does anyone know what other parameters might exist? The 3810 uses the Motorola MPC860 processor which can use 8,16,or 32 bits. I don't know the router uses. > > Thanks! > > John Maybe this is a dumb question, but if some work and some don't, is it just that the ones that don't work are bad? Martin From ian at uax.org.uk Tue Dec 4 07:10:13 2007 From: ian at uax.org.uk (Ian Jolly) Date: Tue, 4 Dec 2007 13:10:13 -0000 Subject: [VoIP] +61 - Another country on CNET Message-ID: <015601c83677$02079a80$0a01a8c0@acer1dd0bbc6d0> Just a quick note to say that I've just had the first call to Australia.. Jack in Adelaide is a collector of old telephones. Unfortunately 'cos there isn't an "Australian" directory page you will not be able to find his number :-) I'll let you know when he is up and running permanently. He currently is experimenting to get his CNET connection up and running in conjunction with his two VoIP connections. The two VoIP circuits are normally on a Netcomm NB9W modem with a Netcomm V200 connected to it. Both have two FXS ports - the interesting feature is that the FXS ports on the NB9W accept pulse dialling !. We tried without success to get an FXS port on the Netcomm V200 set up so Jack then set CNET up instead of the VoIP circuits in the NB9W and we had contact. He's now going to try to get both the CNET and VoIP working at the same time. Hopefully we'll see more folk in Oz joining us. Incidentally if you want to know the time in Adelaide - try http://www.timeanddate.com/worldclock/city.html?n=5 Ian Jolly +44 (0) 352 82 26 (via a 1929 GPO Rural Automatic eXchange!) from CNET - the Heritage Telephone Network +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from Public Telephone Network FWD Telephone No 83 2230 From chad at maine.edu Tue Dec 4 15:49:04 2007 From: chad at maine.edu (Chad Perkins) Date: Tue, 04 Dec 2007 16:49:04 -0500 Subject: [VoIP] Returning CNET numbers In-Reply-To: <020801c82ca5$e3366d00$0201a8c0@Main> Message-ID: <47558500.9176.1BA651C@localhost> On 21 Nov 2007 at 20:20, Paul Wills wrote: > I believe that there were some who proposed that NXX-0002 be a > standard 1000 Hz. tone number. > > I believe Chad Perkin's status monitor depends on it. > PDW I've been quietly listening to this thread waiting to see if some bright light would appear at the end of the tunnel; after a couple weeks I have not seen any. Any body figure out how how we can fix this? The CNET Milliwatt Test depends on getting through to a standard number on each exchange (NXX). 0002 has been a quasi CNET standard for a couple years now. It was the most widely implemented "standard" test number on CNET long before even I joined. One thousand block pooling breaks the CNETMWT script in a pretty significant way. I only see three solutions: 1. Build a stand-alone database of everyones MW, keep up adds, moves, changes (can you say, No) 2. Send Greg all the code, html and scripts to build into the web site (possible, not all too pretty) 3. Evolve to a number plan where thousand block pooling isn't necessary (i.e. predivestiture? North American number plan, kind of like UK folks did). Anyone who has ideas on this is welcome join the conference bridge (+1 700-2663) tonight (6 pm ET) for a detailed explanation / brain storming session in lieu of a lot of replies here on the list. Chad Perkins +1 955-9924 > ----- Original Message ----- > From: "john jones" > To: "Voice Over IP Tandem for Analog Switches" > Sent: Wednesday, November 21, 2007 6:56 PM > Subject: Re: [VoIP] Returning CNET numbers > > > > The idea is for you to pick a number in 594-3xxx and it will be > > marked as a reachability test number. It can be anything you want. > > > > John > > > > ----- Original Message ---- > > From: Paul Wills > > To: Voice Over IP Tandem for Analog Switches > > Sent: Wednesday, November 21, 2007 6:49:43 PM > > Subject: Re: [VoIP] Returning CNET numbers > > > > > > I just thought of something (dangerous though that may be!): > > > > I relinquished everything but the 594-3XXX block of 1000 and just > > realized that my 1000 Hz tone, as per some "standard" is located at > > 594-0002 (as are a few other "special" numbers.) > > > > What will the new standard for the milliwatt tone be since some > > people use it to verify the status of the switch? > > > > PDW > > > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.503 / Virus Database: 269.16.2/1143 - Release Date: > 11/21/07 10:01 AM > From chad at maine.edu Tue Dec 4 15:53:39 2007 From: chad at maine.edu (Chad Perkins) Date: Tue, 04 Dec 2007 16:53:39 -0500 Subject: [VoIP] Returning CNET numbers In-Reply-To: <205631.70688.qm@web34302.mail.mud.yahoo.com> Message-ID: <47558613.19582.1BE9433@localhost> On 21 Nov 2007 at 17:41, john jones wrote: > Yes, but that needs to evolve to deal with 1000 number assignments. > While the concept of using NXX-X002 as the "new" standard, it was felt > that it would be more "friendly" and not very much more difficult to > allow a user to indicate ANY number in their block as a standard test > number. That's a problem. > Greg needs to update the database structure to accommodate this and > Chad will need to update his application as well. > JJ How is the script to know which thousand blocks have been activated? If you are around tonight at 6 o'clock, I could use all the ideas I can get. Chad > ----- Original Message ---- > From: Paul Wills > To: Voice Over IP Tandem for Analog Switches > Sent: Wednesday, November 21, 2007 8:20:35 PM > Subject: Re: [VoIP] Returning CNET numbers > > > I believe that there were some who proposed that NXX-0002 be a > standard > 1000 > Hz. tone number. > > I believe Chad Perkin's status monitor depends on it. > > PDW > > ----- Original Message ----- > From: "john jones" > To: "Voice Over IP Tandem for Analog Switches" > Sent: Wednesday, November 21, 2007 6:56 PM > Subject: Re: [VoIP] Returning CNET numbers > > > > The idea is for you to pick a number in 594-3xxx and it will be > marked as > > a reachability test number. It can be anything you want. > > > > John > > > > ----- Original Message ---- > > From: Paul Wills > > To: Voice Over IP Tandem for Analog Switches > > Sent: Wednesday, November 21, 2007 6:49:43 PM > > Subject: Re: [VoIP] Returning CNET numbers > > > > > > I just thought of something (dangerous though that may be!): > > > > I relinquished everything but the 594-3XXX block of 1000 and just > > realized that my 1000 Hz tone, as per some "standard" is located at > > 594-0002 > (as > > are > > a few other "special" numbers.) > > > > What will the new standard for the milliwatt tone be since some > people > > use > > it to verify the status of the switch? > > > > PDW > > > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.503 / Virus Database: 269.16.2/1143 - Release Date: > 11/21/07 10:01 AM > From jjones3601 at yahoo.com Tue Dec 4 18:46:48 2007 From: jjones3601 at yahoo.com (john jones) Date: Tue, 4 Dec 2007 16:46:48 -0800 (PST) Subject: [VoIP] Returning CNET numbers Message-ID: <981706.91634.qm@web34312.mail.mud.yahoo.com> I'm around now. John 1.687.7700 ----- Original Message ---- From: Chad Perkins To: Voice Over IP Tandem for Analog Switches Sent: Tuesday, December 4, 2007 4:53:39 PM Subject: Re: [VoIP] Returning CNET numbers On 21 Nov 2007 at 17:41, john jones wrote: > Yes, but that needs to evolve to deal with 1000 number assignments. > While the concept of using NXX-X002 as the "new" standard, it was felt > that it would be more "friendly" and not very much more difficult to > allow a user to indicate ANY number in their block as a standard test > number. That's a problem. > Greg needs to update the database structure to accommodate this and > Chad will need to update his application as well. > JJ How is the script to know which thousand blocks have been activated? If you are around tonight at 6 o'clock, I could use all the ideas I can get. Chad > ----- Original Message ---- > From: Paul Wills > To: Voice Over IP Tandem for Analog Switches > Sent: Wednesday, November 21, 2007 8:20:35 PM > Subject: Re: [VoIP] Returning CNET numbers > > > I believe that there were some who proposed that NXX-0002 be a > standard > 1000 > Hz. tone number. > > I believe Chad Perkin's status monitor depends on it. > > PDW > > ----- Original Message ----- > From: "john jones" > To: "Voice Over IP Tandem for Analog Switches" > Sent: Wednesday, November 21, 2007 6:56 PM > Subject: Re: [VoIP] Returning CNET numbers > > > > The idea is for you to pick a number in 594-3xxx and it will be > marked as > > a reachability test number. It can be anything you want. > > > > John > > > > ----- Original Message ---- > > From: Paul Wills > > To: Voice Over IP Tandem for Analog Switches > > Sent: Wednesday, November 21, 2007 6:49:43 PM > > Subject: Re: [VoIP] Returning CNET numbers > > > > > > I just thought of something (dangerous though that may be!): > > > > I relinquished everything but the 594-3XXX block of 1000 and just > > realized that my 1000 Hz tone, as per some "standard" is located at > > 594-0002 > (as > > are > > a few other "special" numbers.) > > > > What will the new standard for the milliwatt tone be since some > people > > use > > it to verify the status of the switch? > > > > PDW > > > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.503 / Virus Database: 269.16.2/1143 - Release Date: > 11/21/07 10:01 AM > _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From greg at vyger.net Tue Dec 4 19:21:52 2007 From: greg at vyger.net (Greg Blakely) Date: Tue, 4 Dec 2007 19:21:52 -0600 Subject: [VoIP] +61 - Another country on CNET Message-ID: He now shows up in http://www.ckts.info/018/ Incidentally, I didn't ask about New Zealand, and have the same question about Australia: Do those countries, under their legacy dial plans, use '0' to denote an out-of-exchange call? Or is that just the UK? > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of Ian Jolly > Sent: Tuesday, December 04, 2007 7:10 AM > To: CNET-UK-I at yahoogroups.com; CNET VoIP > Subject: [VoIP] +61 - Another country on CNET > > Just a quick note to say that I've just had the first call > to Australia.. > > Jack in Adelaide is a collector of old telephones. > Unfortunately 'cos there isn't an "Australian" directory > page you will not be able to find his number :-) I'll let > you know when he is up and running permanently. > > He currently is experimenting to get his CNET connection up > and running in conjunction with his two VoIP connections. The > two VoIP circuits are normally on a Netcomm NB9W modem with > a Netcomm V200 connected to it. Both have two FXS ports - the > interesting feature is that the FXS ports on the NB9W accept > pulse dialling !. We tried without success to get an FXS > port on the Netcomm V200 set up so Jack then set CNET up > instead of the VoIP circuits in the NB9W and we had contact. > He's now going to try to get both the CNET and VoIP working > at the same time. > > Hopefully we'll see more folk in Oz joining us. Incidentally > if you want to know the time in Adelaide - try > http://www.timeanddate.com/worldclock/city.html?n=5 > > Ian Jolly > +44 (0) 352 82 26 (via a 1929 GPO Rural Automatic > eXchange!) from CNET - the Heritage Telephone Network > +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from > +Public Telephone Network > FWD Telephone No 83 2230 > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From ian at uax.org.uk Tue Dec 4 19:46:12 2007 From: ian at uax.org.uk (Ian Jolly) Date: Wed, 5 Dec 2007 01:46:12 -0000 Subject: [VoIP] +61 - Another country on CNET References: Message-ID: <039d01c836e0$9f62b6c0$0a01a8c0@acer1dd0bbc6d0> Tnx Greg Yes both follow the UK '0' for access to the STD calls - See http://www.ummm.net/phonebooks/areacodes/c1971.jpg and http://www.ummm.net/phonebooks/areacodes/d1971.jpg for some of the codes in 1971 - the early days of STD/Distance Dialling. For International - NZ uses 00 plus country code etc. Australia uses 0011 plus country code etc. Hopefully we'll have more from Australia before too long. The guy there is a collector of telephones. He's got a temporary number whilst we do a bit of research into the old STD codes/numbering. I've set a couple of numbers up (21 and 23) for testing on "Cedar Creek exchange" in the mountains to the west of Brisbane. My user name is Bill A Bong Boy - is it hot "here" in the outback by the billabong http://en.wikipedia.org/wiki/Billabong as it approaches the middle of summer - 90 degrees by 9am this morning and expected to reach nearly 100 degrees this afternoon (Thursday) in Adelaide! Ian (on his way back to +44 Land) +44 (0) 352 82 26 (via a 1929 GPO Rural Automatic eXchange!) from CNET - the Heritage Telephone Network +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from Public Telephone Network FWD Telephone No 83 2230 ----- Original Message ----- From: "Greg Blakely" To: "Ian Jolly" ; "Voice Over IP Tandem for Analog Switches" Sent: Wednesday, December 05, 2007 1:21 AM Subject: RE: [VoIP] +61 - Another country on CNET He now shows up in http://www.ckts.info/018/ Incidentally, I didn't ask about New Zealand, and have the same question about Australia: Do those countries, under their legacy dial plans, use '0' to denote an out-of-exchange call? Or is that just the UK? > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of Ian Jolly > Sent: Tuesday, December 04, 2007 7:10 AM > To: CNET-UK-I at yahoogroups.com; CNET VoIP > Subject: [VoIP] +61 - Another country on CNET > > Just a quick note to say that I've just had the first call > to Australia.. > > Jack in Adelaide is a collector of old telephones. > Unfortunately 'cos there isn't an "Australian" directory > page you will not be able to find his number :-) I'll let > you know when he is up and running permanently. > > He currently is experimenting to get his CNET connection up > and running in conjunction with his two VoIP connections. The > two VoIP circuits are normally on a Netcomm NB9W modem with > a Netcomm V200 connected to it. Both have two FXS ports - the > interesting feature is that the FXS ports on the NB9W accept > pulse dialling !. We tried without success to get an FXS > port on the Netcomm V200 set up so Jack then set CNET up > instead of the VoIP circuits in the NB9W and we had contact. > He's now going to try to get both the CNET and VoIP working > at the same time. > > Hopefully we'll see more folk in Oz joining us. Incidentally > if you want to know the time in Adelaide - try > http://www.timeanddate.com/worldclock/city.html?n=5 > > Ian Jolly > +44 (0) 352 82 26 (via a 1929 GPO Rural Automatic > eXchange!) from CNET - the Heritage Telephone Network > +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from > +Public Telephone Network > FWD Telephone No 83 2230 > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.16.13/1169 - Release Date: 03/12/2007 22:56 From ratguy at insightbb.com Wed Dec 5 01:40:29 2007 From: ratguy at insightbb.com (Jayson Smith) Date: Wed, 5 Dec 2007 02:40:29 -0500 Subject: [VoIP] Returning CNET numbers References: <47558500.9176.1BA651C@localhost> Message-ID: <000e01c83712$1becc560$6900a8c0@BOE> Hi, Sorry I wasn't at the conference, I just missed it, didn't even know about it until now. That's what I get for not staying up long enough and checking my E-mail! Anyway, did anything useful come out of it? Jayson ----- Original Message ----- From: "Chad Perkins" To: "Voice Over IP Tandem for Analog Switches" Sent: Tuesday, December 04, 2007 4:49 PM Subject: Re: [VoIP] Returning CNET numbers > On 21 Nov 2007 at 20:20, Paul Wills wrote: > >> I believe that there were some who proposed that NXX-0002 be a >> standard 1000 Hz. tone number. >> >> I believe Chad Perkin's status monitor depends on it. >> PDW > > I've been quietly listening to this thread waiting to see if some bright > light would > appear at the end of the tunnel; after a couple weeks I have not seen any. > > Any body figure out how how we can fix this? The CNET Milliwatt Test > depends on > getting through to a standard number on each exchange (NXX). > > 0002 has been a quasi CNET standard for a couple years now. It was the > most > widely implemented "standard" test number on CNET long before even I > joined. One > thousand block pooling breaks the CNETMWT script in a pretty significant > way. I only > see three solutions: > > 1. Build a stand-alone database of everyones MW, keep up adds, moves, > changes > (can you say, No) > 2. Send Greg all the code, html and scripts to build into the web site > (possible, not all > too pretty) > 3. Evolve to a number plan where thousand block pooling isn't necessary > (i.e. > predivestiture? North American number plan, kind of like UK folks did). > > Anyone who has ideas on this is welcome join the conference bridge (+1 > 700-2663) > tonight (6 pm ET) for a detailed explanation / brain storming session in > lieu of a lot of > replies here on the list. > > Chad Perkins > +1 955-9924 > > >> ----- Original Message ----- >> From: "john jones" >> To: "Voice Over IP Tandem for Analog Switches" >> Sent: Wednesday, November 21, 2007 6:56 PM >> Subject: Re: [VoIP] Returning CNET numbers >> >> >> > The idea is for you to pick a number in 594-3xxx and it will be >> > marked as a reachability test number. It can be anything you want. >> > >> > John >> > >> > ----- Original Message ---- >> > From: Paul Wills >> > To: Voice Over IP Tandem for Analog Switches >> > Sent: Wednesday, November 21, 2007 6:49:43 PM >> > Subject: Re: [VoIP] Returning CNET numbers >> > >> > >> > I just thought of something (dangerous though that may be!): >> > >> > I relinquished everything but the 594-3XXX block of 1000 and just >> > realized that my 1000 Hz tone, as per some "standard" is located at >> > 594-0002 (as are a few other "special" numbers.) >> > >> > What will the new standard for the milliwatt tone be since some >> > people use it to verify the status of the switch? >> > >> > PDW >> > >> > >> > >> >> _______________________________________________ >> VoIP mailing list >> VoIP at ckts.info >> http://lists.ckts.info/mailman/listinfo/voip >> Project Web Page: http://www.ckts.info/ >> >> >> -- >> No virus found in this incoming message. >> Checked by AVG Free Edition. >> Version: 7.5.503 / Virus Database: 269.16.2/1143 - Release Date: >> 11/21/07 10:01 AM >> > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > From hockd at dteenergy.com Wed Dec 5 04:09:04 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Wed, 5 Dec 2007 05:09:04 -0500 Subject: [VoIP] +61 - Another country on CNET In-Reply-To: <015601c83677$02079a80$0a01a8c0@acer1dd0bbc6d0> References: <015601c83677$02079a80$0a01a8c0@acer1dd0bbc6d0> Message-ID: Congrats and welcome Australia! Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: , "CNET VoIP" From: "Ian Jolly" Sent by: voip-bounces at ckts.info Date: 12/04/2007 08:10AM Subject: [VoIP] +61 - Another country on CNET Just a quick note to say that I've just had the first call to Australia.. Jack in Adelaide is a collector of old telephones. Unfortunately 'cos there isn't an "Australian" directory page you will not be able to find his number :-) I'll let you know when he is up and running permanently. He currently is experimenting to get his CNET connection up and running in conjunction with his two VoIP connections. The two VoIP circuits are normally on a Netcomm NB9W modem with a Netcomm V200 connected to it. Both have two FXS ports - the interesting feature is that the FXS ports on the NB9W accept pulse dialling !. We tried without success to get an FXS port on the Netcomm V200 set up so Jack then set CNET up instead of the VoIP circuits in the NB9W and we had contact. He's now going to try to get both the CNET and VoIP working at the same time. Hopefully we'll see more folk in Oz joining us. Incidentally if you want to know the time in Adelaide - try http://www.timeanddate.com/worldclock/city.html?n=5 Ian Jolly +44 (0) 352 82 26 (via a 1929 GPO Rural Automatic eXchange!) from CNET - the Heritage Telephone Network +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from Public Telephone Network FWD Telephone No 83 2230 _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From jnovack at stromberg-carlson.org Wed Dec 5 09:34:18 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Wed, 05 Dec 2007 10:34:18 -0500 Subject: [VoIP] +61 - Another country on CNET In-Reply-To: References: <015601c83677$02079a80$0a01a8c0@acer1dd0bbc6d0> Message-ID: <4756C4FA.4010505@stromberg-carlson.org> I am in communication with another chap in Australia who wants to run AstLinux on a thin client to connect to CNET. He is: Mark McGough I do hope Ian can help him a bit with his dialplan, as I am more country code 1 fluent. In fact, if Ian can send me a sample set of configs off list, I can install before shipping to him AstLinux, though pretty small ( fits in a 64 meg CF card ) works well, and is easy to manage once installed. I used it for several weeks while I was getting around to rebuilding my CNET box after a failing hard drive experience. For those wanting a quick entry into CNET, without the overhead and trouble of AAH and whatever it currently is called, consider AstLinux. I installed it on an HP thin client, but it will also work booting from a pen drive or from a very small hard disk. John Novack Dennis D Hock wrote: > Congrats and welcome Australia! > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: , "CNET VoIP" > From: "Ian Jolly" > Sent by: voip-bounces at ckts.info > Date: 12/04/2007 08:10AM > Subject: [VoIP] +61 - Another country on CNET > > Just a quick note to say that I've just had the first call to Australia.. > > Jack in Adelaide is a collector of old telephones. Unfortunately 'cos > there isn't an "Australian" directory page you will not be able to find > his number :-) I'll let you know when he is up and running permanently. > > He currently is experimenting to get his CNET connection up and running in > conjunction with his two VoIP connections. The two VoIP circuits are > normally on a Netcomm NB9W modem with a Netcomm V200 connected to it. Both > have two FXS ports - the interesting feature is that the FXS ports on the > NB9W accept pulse dialling !. We tried without success to get an FXS port > on the Netcomm V200 set up so Jack then set CNET up instead of the VoIP > circuits in the NB9W and we had contact. He's now going to try to get both > the CNET and VoIP working at the same time. > > Hopefully we'll see more folk in Oz joining us. Incidentally if you want to > know the time in Adelaide - try > http://www.timeanddate.com/worldclock/city.html?n=5 > > Ian Jolly > +44 (0) 352 82 26 (via a 1929 GPO Rural Automatic eXchange!) from > CNET - the Heritage Telephone Network > +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from > Public Telephone Network > FWD Telephone No 83 2230 > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > -- Dog is my co-pilot From ian at uax.org.uk Wed Dec 5 11:51:01 2007 From: ian at uax.org.uk (Ian Jolly) Date: Wed, 5 Dec 2007 17:51:01 -0000 Subject: [VoIP] +61 - Another country on CNET References: <015601c83677$02079a80$0a01a8c0@acer1dd0bbc6d0> <4756C4FA.4010505@stromberg-carlson.org> Message-ID: <016c01c83767$67e3f150$0a01a8c0@acer1dd0bbc6d0> Will do John - I was in touch with Mark a while back when I invited him to join our lists. Ian +61 729868 21 +64 85 32 876 +44 (0) 352 82 26 (via a 1929 GPO Rural Automatic eXchange!) from CNET - the Heritage Telephone Network +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from Public Telephone Network FWD Telephone No 83 2230 ----- Original Message ----- From: John Novack To: Voice Over IP Tandem for Analog Switches Cc: Ian Jolly ; Mark McGough Sent: Wednesday, December 05, 2007 3:34 PM Subject: Re: [VoIP] +61 - Another country on CNET I am in communication with another chap in Australia who wants to run AstLinux on a thin client to connect to CNET. He is: Mark McGough I do hope Ian can help him a bit with his dialplan, as I am more country code 1 fluent. In fact, if Ian can send me a sample set of configs off list, I can install before shipping to him AstLinux, though pretty small ( fits in a 64 meg CF card ) works well, and is easy to manage once installed. I used it for several weeks while I was getting around to rebuilding my CNET box after a failing hard drive experience. For those wanting a quick entry into CNET, without the overhead and trouble of AAH and whatever it currently is called, consider AstLinux. I installed it on an HP thin client, but it will also work booting from a pen drive or from a very small hard disk. John Novack Dennis D Hock wrote: Congrats and welcome Australia! Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: , "CNET VoIP" From: "Ian Jolly" Sent by: voip-bounces at ckts.info Date: 12/04/2007 08:10AM Subject: [VoIP] +61 - Another country on CNET Just a quick note to say that I've just had the first call to Australia.. Jack in Adelaide is a collector of old telephones. Unfortunately 'cos there isn't an "Australian" directory page you will not be able to find his number :-) I'll let you know when he is up and running permanently. He currently is experimenting to get his CNET connection up and running in conjunction with his two VoIP connections. The two VoIP circuits are normally on a Netcomm NB9W modem with a Netcomm V200 connected to it. Both have two FXS ports - the interesting feature is that the FXS ports on the NB9W accept pulse dialling !. We tried without success to get an FXS port on the Netcomm V200 set up so Jack then set CNET up instead of the VoIP circuits in the NB9W and we had contact. He's now going to try to get both the CNET and VoIP working at the same time. Hopefully we'll see more folk in Oz joining us. Incidentally if you want to know the time in Adelaide - try http://www.timeanddate.com/worldclock/city.html?n=5 Ian Jolly +44 (0) 352 82 26 (via a 1929 GPO Rural Automatic eXchange!) from CNET - the Heritage Telephone Network +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from Public Telephone Network FWD Telephone No 83 2230 _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ -- Dog is my co-pilot -- This email has been verified as Virus free Virus Protection and more available at http://www.plus.net ------------------------------------------------------------------------------ No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.16.14/1171 - Release Date: 04/12/2007 19:31 From john_reads_cnet_via_archives at covert.org Wed Dec 5 19:34:38 2007 From: john_reads_cnet_via_archives at covert.org (John R. Covert) Date: Wed, 5 Dec 2007 20:34:38 -0500 (EST) Subject: [VoIP] Timezones... Message-ID: <20071206015032.E2E8D56B74@ns01.ckts.info> If you call my time recordings on CNET 1-263-2525 or 1-263-0123 with (for example) SET(CALLERID(name)=TZ=Australia/Adelaide) you can change the timezone spoken. Uppercase TZ and valid timezone name required. /john From keelan at mail.grenander.com Wed Dec 5 23:07:06 2007 From: keelan at mail.grenander.com (Keelan Lightfoot) Date: Wed, 05 Dec 2007 21:07:06 -0800 Subject: [VoIP] More "Authentic" CNET Message-ID: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> Something that has bothered me about the current direction that CNET is taking is that it is really just a collection of Asterisk servers, with some extensions that happen to point at some old switching equipment. Or recordings of old equipment. It's hard to tell. It is lacking an interactive component, and it isn't really a network in the "classical" sense of the term, where the telephone network existed as a hierarchy of tandem switches. Placing a call meant connecting through multiple switches and trunks to reach your destination. What about the possibility of using a computer's sound card as a 4 wire trunk? Audio could be passed using a simple streaming audio application (something like IHU - ihu.sourceforge.net). Answer supervision/dial pulsing could either be passed as out of band "DC" signals created or received by extra hardware plugged into the server, or as in band SF tones with detection either handled in hardware or software. The interface between sound card and switch could be executed using something like a Zarlink SLIC or COIC to do 2- wire to 4-wire conversion, and handle signaling. Just a thought. - Keelan From Lucky225 at 2600.com Wed Dec 5 23:34:59 2007 From: Lucky225 at 2600.com (Lucky 225) Date: Wed, 5 Dec 2007 23:34:59 -0600 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> Message-ID: I think this just overcomplicates things, however if someone wants to do it all the more power to them. :) On Dec 5, 2007 11:07 PM, Keelan Lightfoot wrote: > Something that has bothered me about the current direction that CNET > is taking is that it is really just a collection of Asterisk servers, > with some extensions that happen to point at some old switching > equipment. Or recordings of old equipment. It's hard to tell. It is > lacking an interactive component, and it isn't really a network in > the "classical" sense of the term, where the telephone network > existed as a hierarchy of tandem switches. Placing a call meant > connecting through multiple switches and trunks to reach your > destination. > > What about the possibility of using a computer's sound card as a 4 > wire trunk? Audio could be passed using a simple streaming audio > application (something like IHU - ihu.sourceforge.net). Answer > supervision/dial pulsing could either be passed as out of band "DC" > signals created or received by extra hardware plugged into the > server, or as in band SF tones with detection either handled in > hardware or software. The interface between sound card and switch > could be executed using something like a Zarlink SLIC or COIC to do 2- > wire to 4-wire conversion, and handle signaling. > > Just a thought. > > - Keelan > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From david at josephson.com Wed Dec 5 23:47:57 2007 From: david at josephson.com (David Josephson) Date: Wed, 05 Dec 2007 21:47:57 -0800 Subject: [VoIP] More "Authentic" CNET In-Reply-To: References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> Message-ID: <47578D0D.9000609@josephson.com> >> existed as a hierarchy of tandem switches. Placing a call meant >> connecting through multiple switches and trunks to reach your >> destination. >> Not necessarily. The critical part of a network is efficient network design -- that Greg's adaptation of DNS to be an ENUM server does elegantly. You design a network according to the costs and benefits at hand -- and we don't pay more for a long distance connection through the public internet than we do for a short one, so there's no point in tandeming through a bunch of switches just to make the >> What about the possibility of using a computer's sound card as a 4 >> wire trunk? Audio could be passed using a simple streaming audio >> application (something like IHU - ihu.sourceforge.net). Answer >> It sounds like you want to reinvent Asterisk. We have drivers for sound cards to do just that, using the g.711 or other coding schemes built in to Asterisk. >> supervision/dial pulsing could either be passed as out of band "DC" >> signals created or received by extra hardware plugged into the >> server, or as in band SF tones with detection either handled in >> hardware or software. The interface between sound card and switch >> could be executed using something like a Zarlink SLIC or COIC to do 2- >> wire to 4-wire conversion, and handle signaling. >> Yes, we have that. What's the purpose? David From madmanmarkau at hotmail.com Wed Dec 5 23:57:18 2007 From: madmanmarkau at hotmail.com (Mad Mark) Date: Thu, 6 Dec 2007 05:57:18 +0000 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> Message-ID: > Date: Wed, 5 Dec 2007 21:07:06 -0800 > From: keelan at mail.grenander.com > To: voip at ckts.info > Subject: [VoIP] More "Authentic" CNET > > Something that has bothered me about the current direction that CNET > is taking is that it is really just a collection of Asterisk servers, > with some extensions that happen to point at some old switching > equipment. Or recordings of old equipment. It's hard to tell. It is > lacking an interactive component, and it isn't really a network in > the "classical" sense of the term, where the telephone network > existed as a hierarchy of tandem switches. Placing a call meant > connecting through multiple switches and trunks to reach your > destination. To tell you the truth, I would love to have something like this set up. However, over VoIP, the latency would become an absolute killer, and some exchanges could have their connections saturated. Not to mention administering a managed routing scheme including alternate routing/homing. Some switches are not online 24/7 as far as I've seen. > > What about the possibility of using a computer's sound card as a 4 > wire trunk? Audio could be passed using a simple streaming audio > application (something like IHU - ihu.sourceforge.net). Answer > supervision/dial pulsing could either be passed as out of band "DC" > signals created or received by extra hardware plugged into the > server, or as in band SF tones with detection either handled in > hardware or software. The interface between sound card and switch > could be executed using something like a Zarlink SLIC or COIC to do 2- > wire to 4-wire conversion, and handle signaling. Nice idea for sure, but I think it would be better to get some switching hardware with E&M lead input/output. (I think it's E&M that's used for bidirectional 2-wire trunks... Can anyone confirm?) > > Just a thought. Keep up the good work :) > > - Keelan > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ _________________________________________________________________ What are you waiting for? Join Lavalife FREE http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Flavalife9%2Eninemsn%2Ecom%2Eau%2Fclickthru%2Fclickthru%2Eact%3Fid%3Dninemsn%26context%3Dan99%26locale%3Den%5FAU%26a%3D30288&_t=764581033&_r=email_taglines_Join_free_OCT07&_m=EXT From keelan at mail.grenander.com Thu Dec 6 00:44:12 2007 From: keelan at mail.grenander.com (Keelan Lightfoot) Date: Wed, 05 Dec 2007 22:44:12 -0800 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <47578D0D.9000609@josephson.com> References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> <47578D0D.9000609@josephson.com> Message-ID: <1B5C8961-7BA2-41E8-A0BD-BB12CBA67F4C@mail.grenander.com> > It sounds like you want to reinvent Asterisk. We have drivers for > sound > cards to do just that, using the g.711 or other coding schemes > built in > to Asterisk. The "problem" is that Asterisk passes abstract signaling information between nodes; "so-and-so dialed 1235, the person at 12345 is busy, etc." What I'm suggesting is a method of passing very basic state information (virtual E&M leads) between collectors' switches as though they were connected via wire trunks (or carrier). If I dial one of the extensions on a collectors step switch connected to CNET, I don't hear the switch until asterisk has finished pulsing out the dialed digits. I may as well be listening to a recording. some members have even gone so far as to inject canned RP noise into the connection, just to make things a bit more lively (I suppose). >>> supervision/dial pulsing could either be passed as out of band "DC" >>> signals created or received by extra hardware plugged into the >>> server, or as in band SF tones with detection either handled in >>> hardware or software. The interface between sound card and switch >>> could be executed using something like a Zarlink SLIC or COIC to >>> do 2- >>> wire to 4-wire conversion, and handle signaling. >>> > Yes, we have that. What's the purpose? I was listening to the radio a couple days ago, and a couple commentators were critiquing TV christmas specials. One of the commentators brought up "It's a Wonderful Life", and argued against it on the grounds that it was based on an impractical premise, that it was based fantasy and lacked realism. The other commentator stated something along the lines of "This is a holiday where one of the main events is fat man squeezing down a chimney, to give billions of people gifts in the short span of an evening, and you want to talk about practicality and reality?" The purpose would be more of an academic exercise. The Asterisk CNET is sterile and predictable. Once you've run through the list of numbers and listened to everyone's ring-no-answer, asterisk milliwatt, or busy tone lines, you've pretty much played it out. If we were concerned about practicality or purpose, we would sell our switches for scrap and invest the few dollars earned in a mutual fund. The spare space in our basements could be used for more practical things like storage. - Keelan From keelan at mail.grenander.com Thu Dec 6 00:56:31 2007 From: keelan at mail.grenander.com (Keelan Lightfoot) Date: Wed, 05 Dec 2007 22:56:31 -0800 Subject: [VoIP] More "Authentic" CNET In-Reply-To: References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> Message-ID: <783DCC74-F5F5-422B-8D0B-DEB0A204EA66@mail.grenander.com> > To tell you the truth, I would love to have something like this set > up. > However, over VoIP, the latency would become an absolute killer, > and some > exchanges could have their connections saturated. Not to mention > administering > a managed routing scheme including alternate routing/homing. Some > switches > are not online 24/7 as far as I've seen. I think this would add to the liveliness of CNET. This is a network of boat anchors -- getting an ATB condition would not mean the end of the world. Administration would be difficult, but I think central administration wouldn't be necessary; individual collectors could get together and sort out who they are going to peer with. This may exclude the possibility of a uniform dialing plan, but wouldn't that add to the depth of the network? > Nice idea for sure, but I think it would be better to get some > switching > hardware with E&M lead input/output. (I think it's E&M that's used for > bidirectional 2-wire trunks... Can anyone confirm?) E&M would be nice for the switches that supported it, but for many switches, E&M trunking equipment is probably very difficult to come by. It would make the hardware implementation much simpler, though. From duncan.b.smith at gmail.com Thu Dec 6 01:16:50 2007 From: duncan.b.smith at gmail.com (Duncan Smith) Date: Wed, 5 Dec 2007 23:16:50 -0800 Subject: [VoIP] Good source for Channel Bank Cards et cetera Message-ID: <20071206071650.GI18373@5.7.5.5.6.6.6.6.0.2.1.e164.arpa> I was talking with Phil on Tuesday, and I happened to mention that I have bought several (twelve) D4 channel bank cards from a particular reseller, and have been much pleased with their service. So, I'd like to widen my recommendation. Bell Enterprise () has an excellent selection of channel bank cards. They also have lots of other items; take your favorite J-number and drop it into their search engine. Here's a link to their list of D4 parts available: http://www.bell-enterprise.com/parts/lucent/d4/ Bell Enterprise packs well, has decent prices (apparently $10 for a DPT D4 card is rather low), and they offer a one-year warranty on everything you purchase from them. I have no relationship with Bell Enterprise, except that I'm a satisfied customer. :) -- Duncan Smith --------\ http://students.washington.edu/f/ /--- () ascii ribbon \--- Signed/encrypted mail preferred ---/ /\ campaign [ against html mail ] [ support open formats ] From madmanmarkau at hotmail.com Thu Dec 6 01:17:16 2007 From: madmanmarkau at hotmail.com (Mad Mark) Date: Thu, 6 Dec 2007 07:17:16 +0000 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <783DCC74-F5F5-422B-8D0B-DEB0A204EA66@mail.grenander.com> References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> <783DCC74-F5F5-422B-8D0B-DEB0A204EA66@mail.grenander.com> Message-ID: > Date: Wed, 5 Dec 2007 22:56:31 -0800 > From: keelan at mail.grenander.com > To: voip at ckts.info > Subject: Re: [VoIP] More "Authentic" CNET > > > To tell you the truth, I would love to have something like this set > > up. > > However, over VoIP, the latency would become an absolute killer, > > and some > > exchanges could have their connections saturated. Not to mention > > administering > > a managed routing scheme including alternate routing/homing. Some > > switches > > are not online 24/7 as far as I've seen. > > I think this would add to the liveliness of CNET. This is a network > of boat anchors -- getting an ATB condition would not mean the end of > the world. Administration would be difficult, but I think central > administration wouldn't be necessary; individual collectors could get > together and sort out who they are going to peer with. This may > exclude the possibility of a uniform dialing plan, but wouldn't that > add to the depth of the network? That it would, for sure. However, with the old phone system, they did have some dialplans, homing plans etc... You're talking about an emulation of the old phone network, right to the level of inter-office trunk groups. Managed homing is a *must* IMHO. I've only really ever studied the American phone system (eternal thanks to Evan Doorbell for the opportunity) so I'm no authority in other countries. I don't really believe in the ad-hoc homing method you've outlined. I can just see a small handful of switches being maxed out routing calls. There's also the issue of the latency of VoIP that hasn't been addressed. This could potentially be a major drawback. I've stacked calls over VoIP/CNet, and using only 2 or three links the lag-times were in the order of 4-5 seconds, making conversation nigh on impossible. > > > Nice idea for sure, but I think it would be better to get some > > switching > > hardware with E&M lead input/output. (I think it's E&M that's used for > > bidirectional 2-wire trunks... Can anyone confirm?) > > E&M would be nice for the switches that supported it, but for many > switches, E&M trunking equipment is probably very difficult to come > by. It would make the hardware implementation much simpler, though. If only I knew the inns and outs of E&M and other trunking technologies. _________________________________________________________________ It's simple! Sell your car for just $30 at CarPoint.com.au http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fsecure%2Dau%2Eimrworldwide%2Ecom%2Fcgi%2Dbin%2Fa%2Fci%5F450304%2Fet%5F2%2Fcg%5F801459%2Fpi%5F1004813%2Fai%5F859641&_t=762955845&_r=tig_OCT07&_m=EXT From stfkerman at jps.net Thu Dec 6 02:37:30 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 03:37:30 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> <783DCC74-F5F5-422B-8D0B-DEB0A204EA66@mail.grenander.com> Message-ID: <4757B4CA.3070408@jps.net> Mad Mark wrote: >> Date: Wed, 5 Dec 2007 22:56:31 -0800 >> From: keelan at mail.grenander.com >> To: voip at ckts.info >> Subject: Re: [VoIP] More "Authentic" CNET >>> Nice idea for sure, but I think it would be better to get some >>> switching hardware with E&M lead input/output. (I think it's E&M >>> that's used for bidirectional 2-wire trunks... Can anyone confirm?) >> E&M would be nice for the switches that supported it, but for many >> switches, E&M trunking equipment is probably very difficult to come >> by. It would make the hardware implementation much simpler, though. > If only I knew the inns and outs of E&M and other trunking technologies. E&M trunks were normally used on derived facilities: very long metallic trunks that required polar duplex (DX) signaling because of the conductor resistance and usually 4 wire transmission because of the VF loss, and also used many "carrier" (analog FDM or digital PCM) trunks. Prior to widespread use of T1 carrier typical metallic interoffice trunks used "loop reverse battery supervision". Even many T1 trunks used loop signaling too. The widely used DPO and DPT T1 channel cards are for loop/reverse battery signaling trunks. 2600 Hz "E" signaling units were also used for signaling over many carrier derived trunks. Many of these, particularly the 2 wire ones, used loop reverse battery signaling too. Loop signaling was most commonly used on 1-way trunks. E&M was more the rule for 2-way 4-wire toll trunks. Loop/reverse battery refers to the methods of controlling seizure and release towards the terminating end and answer supervision back to towards the originating end. An older loop method used in manual common battery practice was loop high/low where answer supervision was indicated by drawing or delivering greater current. This method was also used in some PBX tie trunk circuits that might be found in some equipment collectors' systems. SXS and most other electromechnical switching systems use loop supervision internally and are inherently compatible with loop supervision trunks without additional trunk equipment as such. FXS and FXO channel interfaces inherently use loop supervision because they emulate a telephone set, which uses loop supervision. So using E&M would be a complication all around. Steph From madmanmarkau at hotmail.com Thu Dec 6 04:00:42 2007 From: madmanmarkau at hotmail.com (Mad Mark) Date: Thu, 6 Dec 2007 10:00:42 +0000 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <4757B4CA.3070408@jps.net> References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> <783DCC74-F5F5-422B-8D0B-DEB0A204EA66@mail.grenander.com> <4757B4CA.3070408@jps.net> Message-ID: And now I *do* know the inns and outs of E&M :) Cheers, that'll go in my notes folder. > Date: Thu, 6 Dec 2007 03:37:30 -0500 > From: stfkerman at jps.net > To: voip at ckts.info > Subject: Re: [VoIP] More "Authentic" CNET > > > > Mad Mark wrote: > >> Date: Wed, 5 Dec 2007 22:56:31 -0800 > >> From: keelan at mail.grenander.com > >> To: voip at ckts.info > >> Subject: Re: [VoIP] More "Authentic" CNET > >>> Nice idea for sure, but I think it would be better to get some > >>> switching hardware with E&M lead input/output. (I think it's E&M > >>> that's used for bidirectional 2-wire trunks... Can anyone confirm?) > >> E&M would be nice for the switches that supported it, but for many > >> switches, E&M trunking equipment is probably very difficult to come > >> by. It would make the hardware implementation much simpler, though. > > If only I knew the inns and outs of E&M and other trunking technologies. > E&M trunks were normally used on derived facilities: very long metallic > trunks that required polar duplex (DX) signaling because of the > conductor resistance and usually 4 wire transmission because of the VF > loss, and also used many "carrier" (analog FDM or digital PCM) trunks. > Prior to widespread use of T1 carrier typical metallic interoffice > trunks used "loop reverse battery supervision". > > Even many T1 trunks used loop signaling too. The widely used DPO and > DPT T1 channel cards are for loop/reverse battery signaling trunks. > 2600 Hz "E" signaling units were also used for signaling over many > carrier derived trunks. Many of these, particularly the 2 wire ones, > used loop reverse battery signaling too. Loop signaling was most > commonly used on 1-way trunks. E&M was more the rule for 2-way 4-wire > toll trunks. > > Loop/reverse battery refers to the methods of controlling seizure and > release towards the terminating end and answer supervision back to > towards the originating end. An older loop method used in manual common > battery practice was loop high/low where answer supervision was > indicated by drawing or delivering greater current. This method was > also used in some PBX tie trunk circuits that might be found in some > equipment collectors' systems. > > SXS and most other electromechnical switching systems use loop > supervision internally and are inherently compatible with loop > supervision trunks without additional trunk equipment as such. FXS and > FXO channel interfaces inherently use loop supervision because they > emulate a telephone set, which uses loop supervision. So using E&M > would be a complication all around. > > Steph > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ _________________________________________________________________ Overpaid or Underpaid? Check our comprehensive Salary Centre http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fcontent%2Emycareer%2Ecom%2Eau%2Fsalary%2Dcentre%3Fs%5Fcid%3D595810&_t=766724125&_r=Hotmail_Email_Tagline_MyCareer_Oct07&_m=EXT From hockd at dteenergy.com Thu Dec 6 04:12:32 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Thu, 6 Dec 2007 05:12:32 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <1B5C8961-7BA2-41E8-A0BD-BB12CBA67F4C@mail.grenander.com> References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> <47578D0D.9000609@josephson.com>, <1B5C8961-7BA2-41E8-A0BD-BB12CBA67F4C@mail.grenander.com> Message-ID: Well said Keelan. Some might argue that CNET needs to be able to replicate the full gamut of the original experience, while we seem to be forgetting just how far we have all collectively managed to come in the span of a few short years (2 going on 3 by my count). Given time and motivation I think the experience will most certainly continue to improve and be added to. It may be that now that we are able to link to each other we have reached a new conundrum in that what do we talk about and who can we talk to? This is not much different in my opinion than any other hobby in which much of the joy and fun comes in the construction. A case in point being the amatuer radio operator who linking with some one in another country chats about the weather and the equipment they each have or the model railroader who builds mutltiple layouts or continues to build that layout that is never quite finished. Some of the fun is in the building. One last thing as we all have varying memories and fantasies of what the network is, was, should have been , could have been, the main point I think is what other commercial company has managed to expand a network so quickly onto the international scene as this group has? After all this is all being done as a labor of love by many who have varying interests in the many aspects this offers us. I think we should set back an=t this time of year and reflect on where we are and how far we have come. Just my two cents. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: Keelan Lightfoot Sent by: voip-bounces at ckts.info Date: 12/06/2007 01:44AM Subject: Re: [VoIP] More "Authentic" CNET > It sounds like you want to reinvent Asterisk. We have drivers for > sound > cards to do just that, using the g.711 or other coding schemes > built in > to Asterisk. The "problem" is that Asterisk passes abstract signaling information between nodes; "so-and-so dialed 1235, the person at 12345 is busy, etc." What I'm suggesting is a method of passing very basic state information (virtual E&M leads) between collectors' switches as though they were connected via wire trunks (or carrier). If I dial one of the extensions on a collectors step switch connected to CNET, I don't hear the switch until asterisk has finished pulsing out the dialed digits. I may as well be listening to a recording. some members have even gone so far as to inject canned RP noise into the connection, just to make things a bit more lively (I suppose). >>> supervision/dial pulsing could either be passed as out of band "DC" >>> signals created or received by extra hardware plugged into the >>> server, or as in band SF tones with detection either handled in >>> hardware or software. The interface between sound card and switch >>> could be executed using something like a Zarlink SLIC or COIC to >>> do 2- >>> wire to 4-wire conversion, and handle signaling. >>> > Yes, we have that. What's the purpose? I was listening to the radio a couple days ago, and a couple commentators were critiquing TV christmas specials. One of the commentators brought up "It's a Wonderful Life", and argued against it on the grounds that it was based on an impractical premise, that it was based fantasy and lacked realism. The other commentator stated something along the lines of "This is a holiday where one of the main events is fat man squeezing down a chimney, to give billions of people gifts in the short span of an evening, and you want to talk about practicality and reality?" The purpose would be more of an academic exercise. The Asterisk CNET is sterile and predictable. Once you've run through the list of numbers and listened to everyone's ring-no-answer, asterisk milliwatt, or busy tone lines, you've pretty much played it out. If we were concerned about practicality or purpose, we would sell our switches for scrap and invest the few dollars earned in a mutual fund. The spare space in our basements could be used for more practical things like storage. - Keelan _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From ratguy at insightbb.com Thu Dec 6 04:36:42 2007 From: ratguy at insightbb.com (Jayson Smith) Date: Thu, 6 Dec 2007 05:36:42 -0500 Subject: [VoIP] More "Authentic" CNET References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> <47578D0D.9000609@josephson.com> <1B5C8961-7BA2-41E8-A0BD-BB12CBA67F4C@mail.grenander.com> Message-ID: <000701c837f3$e416f6f0$6900a8c0@BOE> Agreed, Dennis. I think to get a truly authentic CNET, we'd have to reinvent the old analog telephone network. The problem, as I see it, is that the Internet is, by its very nature, a digital medium, while what we're wanting to simulate is an analog medium. Imho, any attempt to simulate authentic telephone networks over the Internet would be just that, a simulation. You wouldn't get any unpredictable analog things like crosstalk, signaling crosstalk, other cool noises, etc. Even if those things were deliberately built into the network, we'd still know they were deliberate, and not a natural occurrence. On the subject of dial pulsing noise, Doug, office code 366, is one person who does allow dial pulsing noise to pass back to the originating caller. And as for canned RP noise, there is an RP simulator which uses bits and pieces from an Evan Doorbell recording. So, while it's technically canned sound, since the noises themselves are not being generated on the fly, the sequence of the noises is being determined on the fly, and in my view, that makes it not totally canned. Jayson ----- Original Message ----- From: "Dennis D Hock" To: "Voice Over IP Tandem for Analog Switches" Cc: "Voice Over IP Tandem for Analog Switches" Sent: Thursday, December 06, 2007 5:12 AM Subject: Re: [VoIP] More "Authentic" CNET > > Well said Keelan. > > Some might argue that CNET needs to be able to replicate the full gamut of > the original experience, while we seem to be forgetting just how far we > have all collectively managed to come in the span of a few short years (2 > going on 3 by my count). Given time and motivation I think the experience > will most certainly continue to improve and be added to. > > It may be that now that we are able to link to each other we have reached > a > new conundrum in that what do we talk about and who can we talk to? This > is > not much different in my opinion than any other hobby in which much of the > joy and fun comes in the construction. A case in point being the amatuer > radio operator who linking with some one in another country chats about > the > weather and the equipment they each have or the model railroader who > builds > mutltiple layouts or continues to build that layout that is never quite > finished. Some of the fun is in the building. > > One last thing as we all have varying memories and fantasies of what the > network is, was, should have been , could have been, the main point I > think > is what other commercial company has managed to expand a network so > quickly > onto the international scene as this group has? After all this is all > being done as a labor of love by many who have varying interests in the > many aspects this offers us. > > I think we should set back an=t this time of year and reflect on where we > are and how far we have come. > > Just my two cents. > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: Keelan Lightfoot > Sent by: voip-bounces at ckts.info > Date: 12/06/2007 01:44AM > Subject: Re: [VoIP] More "Authentic" CNET > >> It sounds like you want to reinvent Asterisk. We have drivers for >> sound >> cards to do just that, using the g.711 or other coding schemes >> built in >> to Asterisk. > > The "problem" is that Asterisk passes abstract signaling information > between nodes; "so-and-so dialed 1235, the person at 12345 is busy, > etc." What I'm suggesting is a method of passing very basic state > information (virtual E&M leads) between collectors' switches as > though they were connected via wire trunks (or carrier). If I dial > one of the extensions on a collectors step switch connected to CNET, > I don't hear the switch until asterisk has finished pulsing out the > dialed digits. I may as well be listening to a recording. some > members have even gone so far as to inject canned RP noise into the > connection, just to make things a bit more lively (I suppose). > >>>> supervision/dial pulsing could either be passed as out of band "DC" >>>> signals created or received by extra hardware plugged into the >>>> server, or as in band SF tones with detection either handled in >>>> hardware or software. The interface between sound card and switch >>>> could be executed using something like a Zarlink SLIC or COIC to >>>> do 2- >>>> wire to 4-wire conversion, and handle signaling. >>>> >> Yes, we have that. What's the purpose? > > I was listening to the radio a couple days ago, and a couple > commentators were critiquing TV christmas specials. One of the > commentators brought up "It's a Wonderful Life", and argued against > it on the grounds that it was based on an impractical premise, that > it was based fantasy and lacked realism. The other commentator stated > something along the lines of "This is a holiday where one of the main > events is fat man squeezing down a chimney, to give billions of > people gifts in the short span of an evening, and you want to talk > about practicality and reality?" > > The purpose would be more of an academic exercise. The Asterisk CNET > is sterile and predictable. Once you've run through the list of > numbers and listened to everyone's ring-no-answer, asterisk > milliwatt, or busy tone lines, you've pretty much played it out. > > If we were concerned about practicality or purpose, we would sell our > switches for scrap and invest the few dollars earned in a mutual > fund. The spare space in our basements could be used for more > practical things like storage. > > - Keelan > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > From ratguy at insightbb.com Thu Dec 6 04:53:03 2007 From: ratguy at insightbb.com (Jayson Smith) Date: Thu, 6 Dec 2007 05:53:03 -0500 Subject: [VoIP] More "Authentic" CNET References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> <47578D0D.9000609@josephson.com> <1B5C8961-7BA2-41E8-A0BD-BB12CBA67F4C@mail.grenander.com> Message-ID: <000301c837f6$2d20b5a0$6900a8c0@BOE> On the subject of a fat man hopping down chimneys and delivering presents all around the world, a few years ago I found a humorous tidbit I thought I'd share. I know this is totally off-topic, but who cares? For what it's worth, as a kid, I seem to remember thinking that Santa got around some of these physical limitations by there actually being several Santas, each of which handled a different region of the country. I guess I wasn't thinking about the rest of the world. Funny what kids think, huh? I also had it in my head that, if Santa brought you stuff, and you didn't get up early enough on Christmas morning, he'd come back and take your stuff away, since you obviously had no interest in whatever he had brought. Jayson. Santa Facts Subject: Happy Holidays! From: Randy Kuehn Date: Tue, 20 Dec 1994 02:26:15 -0600 SOME FACTS ABOUT SANTA CLAUS 1) No known species of reindeer can fly. But there are 300,000 species of living organisms yet to be classified, and while most of these are insects and germs, this does not completely rule out flying reindeer which only Santa has ever seen. 2) There are 2 billion children in the world (persons under 18). But since Santa doesn't (appear) to handle Muslim, Hindu, Jewish, or Buddhist children, that reduces the workload by 85% of the total - leaving 378 million according to the Population Reference Bureau. At an average (census) rate of 3.5 children per household, that's 91.8 million homes. One presumes there is at least one good child per house. 3) Santa has 31 hours of Christmas to work with, thanks to the different time zones and the rotation of the earth, assuming he travels east to west (which seems logical). This works out to 822.6 visits per second. This is to say that for each Christian household with good children, Santa has 1/1000 the of a second to park, hop out of the sleigh, jump down the chimney, fill the stocking, distribute the remaining presents under the tree, eat whatever snacks have been left, get back up the chimney, get back into the sleigh and move on to the next house. Assuming that each of these 91.8 million stops are evenly distributed around the earth (which, of course, we know to be false but for the purposes of our calculations we will accept), we are now talking about 0.78 miles per household, a total trip of 75.5 million miles, not counting stops to do what most of us do at least once every 31 hours, plus feeding, etc. That means that Santa's sleigh is moving at 650 miles per second, 3000 times the speed of sound. For purposes of comparison, the fastest man-made vehicle on earth, the Ulysses space probe, moves at a poky 27.4 miles per second - a conventional reindeer can run, at tops 25-30 miles per hour. 4) The payload on the sleigh adds another interesting element. Assuming each child gets nothing more then a medium sized LEGO set (2 lbs), the sleigh is carrying 321300 tons, not counting Santa, who is invariably described as overweight. On land, conventional reindeer can pull no more than 300 pounds. Even granting the 'flying reindeer' can pull TEN TIMES that normal amount, we cannot do the job with eight, or even nine - we need 214200 reindeer. This increased the payload - not even counting the weight of the sleigh to 353430 tons. Again for comparison, this is four timed the weight of the HMS Queen Elizabeth. 5) 353000 tons travelling at 650 miles per second creates enormous air resistance. This will heat the reindeer up in the same fashion as spacecrafts re-entering the earth's atmosphere. The lead pair will absorb 14.3 QUINTILLION joules of energy per second, each. In short, they will burst into flames almost instantaneously, exposing the reindeer behind them, and creating a deafening sonic boom in their wake. The entire reindeer team will be vaporized within 4.26 thousandths of a second. Santa meanwhile, will be subject to centrifugal forces of 17500.06 times greater than gravity. A 250 lb Santa (which seems ludicrously slim) would be pinned to the back of the sleigh by a 4,315,015 pound force. In conclusion, if Santa ever DID deliver presents on Christmas eve, he's now dead. From hockd at dteenergy.com Thu Dec 6 06:02:28 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Thu, 6 Dec 2007 07:02:28 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <000301c837f6$2d20b5a0$6900a8c0@BOE> References: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> <47578D0D.9000609@josephson.com> <1B5C8961-7BA2-41E8-A0BD-BB12CBA67F4C@mail.grenander.com>, <000301c837f6$2d20b5a0$6900a8c0@BOE> Message-ID: Nonsense! As we all know Santa is magical and can transcend the laws of Physics! ;-) Dennis -----voip-bounces at ckts.info wrote: ----- To: "Voice Over IP Tandem for Analog Switches" From: "Jayson Smith" Sent by: voip-bounces at ckts.info Date: 12/06/2007 05:53AM Subject: Re: [VoIP] More "Authentic" CNET On the subject of a fat man hopping down chimneys and delivering presents all around the world, a few years ago I found a humorous tidbit I thought I'd share. I know this is totally off-topic, but who cares? For what it's worth, as a kid, I seem to remember thinking that Santa got around some of these physical limitations by there actually being several Santas, each of which handled a different region of the country. I guess I wasn't thinking about the rest of the world. Funny what kids think, huh? I also had it in my head that, if Santa brought you stuff, and you didn't get up early enough on Christmas morning, he'd come back and take your stuff away, since you obviously had no interest in whatever he had brought. Jayson. Santa Facts Subject: Happy Holidays! From: Randy Kuehn Date: Tue, 20 Dec 1994 02:26:15 -0600 SOME FACTS ABOUT SANTA CLAUS 1) No known species of reindeer can fly. But there are 300,000 species of living organisms yet to be classified, and while most of these are insects and germs, this does not completely rule out flying reindeer which only Santa has ever seen. 2) There are 2 billion children in the world (persons under 18). But since Santa doesn't (appear) to handle Muslim, Hindu, Jewish, or Buddhist children, that reduces the workload by 85% of the total - leaving 378 million according to the Population Reference Bureau. At an average (census) rate of 3.5 children per household, that's 91.8 million homes. One presumes there is at least one good child per house. 3) Santa has 31 hours of Christmas to work with, thanks to the different time zones and the rotation of the earth, assuming he travels east to west (which seems logical). This works out to 822.6 visits per second. This is to say that for each Christian household with good children, Santa has 1/1000 the of a second to park, hop out of the sleigh, jump down the chimney, fill the stocking, distribute the remaining presents under the tree, eat whatever snacks have been left, get back up the chimney, get back into the sleigh and move on to the next house. Assuming that each of these 91.8 million stops are evenly distributed around the earth (which, of course, we know to be false but for the purposes of our calculations we will accept), we are now talking about 0.78 miles per household, a total trip of 75.5 million miles, not counting stops to do what most of us do at least once every 31 hours, plus feeding, etc. That means that Santa's sleigh is moving at 650 miles per second, 3000 times the speed of sound. For purposes of comparison, the fastest man-made vehicle on earth, the Ulysses space probe, moves at a poky 27.4 miles per second - a conventional reindeer can run, at tops 25-30 miles per hour. 4) The payload on the sleigh adds another interesting element. Assuming each child gets nothing more then a medium sized LEGO set (2 lbs), the sleigh is carrying 321300 tons, not counting Santa, who is invariably described as overweight. On land, conventional reindeer can pull no more than 300 pounds. Even granting the 'flying reindeer' can pull TEN TIMES that normal amount, we cannot do the job with eight, or even nine - we need 214200 reindeer. This increased the payload - not even counting the weight of the sleigh to 353430 tons. Again for comparison, this is four timed the weight of the HMS Queen Elizabeth. 5) 353000 tons travelling at 650 miles per second creates enormous air resistance. This will heat the reindeer up in the same fashion as spacecrafts re-entering the earth's atmosphere. The lead pair will absorb 14.3 QUINTILLION joules of energy per second, each. In short, they will burst into flames almost instantaneously, exposing the reindeer behind them, and creating a deafening sonic boom in their wake. The entire reindeer team will be vaporized within 4.26 thousandths of a second. Santa meanwhile, will be subject to centrifugal forces of 17500.06 times greater than gravity. A 250 lb Santa (which seems ludicrously slim) would be pinned to the back of the sleigh by a 4,315,015 pound force. In conclusion, if Santa ever DID deliver presents on Christmas eve, he's now dead. _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From hockd at dteenergy.com Thu Dec 6 06:07:20 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Thu, 6 Dec 2007 07:07:20 -0500 Subject: [VoIP] Returning CNET numbers In-Reply-To: <47558500.9176.1BA651C@localhost> References: <47558500.9176.1BA651C@localhost> Message-ID: Chad as I am just getting to your message I take it I missed the conference call. Can you send me a brief synopsis if you had a decent turnout for discussion. If you needed to reschedule can you offer maybe two days notice as it is sometimes difficult to for me to wade through all the email I receive while working. Thank you, Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: "Chad Perkins" Sent by: voip-bounces at ckts.info Date: 12/04/2007 04:49PM Subject: Re: [VoIP] Returning CNET numbers On 21 Nov 2007 at 20:20, Paul Wills wrote: > I believe that there were some who proposed that NXX-0002 be a > standard 1000 Hz. tone number. > > I believe Chad Perkin's status monitor depends on it. > PDW I've been quietly listening to this thread waiting to see if some bright light would appear at the end of the tunnel; after a couple weeks I have not seen any. Any body figure out how how we can fix this? The CNET Milliwatt Test depends on getting through to a standard number on each exchange (NXX). 0002 has been a quasi CNET standard for a couple years now. It was the most widely implemented "standard" test number on CNET long before even I joined. One thousand block pooling breaks the CNETMWT script in a pretty significant way. I only see three solutions: 1. Build a stand-alone database of everyones MW, keep up adds, moves, changes (can you say, No) 2. Send Greg all the code, html and scripts to build into the web site (possible, not all too pretty) 3. Evolve to a number plan where thousand block pooling isn't necessary (i.e. predivestiture? North American number plan, kind of like UK folks did). Anyone who has ideas on this is welcome join the conference bridge (+1 700-2663) tonight (6 pm ET) for a detailed explanation / brain storming session in lieu of a lot of replies here on the list. Chad Perkins +1 955-9924 > ----- Original Message ----- > From: "john jones" > To: "Voice Over IP Tandem for Analog Switches" > Sent: Wednesday, November 21, 2007 6:56 PM > Subject: Re: [VoIP] Returning CNET numbers > > > > The idea is for you to pick a number in 594-3xxx and it will be > > marked as a reachability test number. It can be anything you want. > > > > John > > > > ----- Original Message ---- > > From: Paul Wills > > To: Voice Over IP Tandem for Analog Switches > > Sent: Wednesday, November 21, 2007 6:49:43 PM > > Subject: Re: [VoIP] Returning CNET numbers > > > > > > I just thought of something (dangerous though that may be!): > > > > I relinquished everything but the 594-3XXX block of 1000 and just > > realized that my 1000 Hz tone, as per some "standard" is located at > > 594-0002 (as are a few other "special" numbers.) > > > > What will the new standard for the milliwatt tone be since some > > people use it to verify the status of the switch? > > > > PDW > > > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.503 / Virus Database: 269.16.2/1143 - Release Date: > 11/21/07 10:01 AM > _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From ratguy at insightbb.com Thu Dec 6 07:35:22 2007 From: ratguy at insightbb.com (Jayson Smith) Date: Thu, 6 Dec 2007 08:35:22 -0500 Subject: [VoIP] Time and temp, weather, etc. Message-ID: <002501c8380c$d9d09e30$6900a8c0@BOE> Hi, I think it would be cool if some people, myself included, could provide real-time time and temperature, and possibly weather, service, without having to place a call to a local number, since this call might cost money if placed through a voip provider. Has anyone ever thought of designing something like a thin client running Asterisk, Astlinux or whatever, with a time and temperature app? How would the thin client receive the temperature data? I actually do have a talking weather station with a wireless probe for receiving temperature and relative humidity readings. Unfortunately, that's about all that talks, other than a very short twelve to twenty-four hour weather forecast based on barometric pressure. And as far as I know, there are no audio outputs or anything, so it's just a stand-alone device. Shane, I know you once offered to build a server with a time and temp application for anyone who could supply recordings. How are you receiving temperature data for your own T&T number? Has anyone ever recorded Alan's or anyone else's T&T machine with Jane Barbe or similar announcements? Just a few thoughts. Jayson From ikj1234i at yahoo.com Thu Dec 6 09:10:16 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Thu, 6 Dec 2007 07:10:16 -0800 (PST) Subject: [VoIP] More "Authentic" CNET In-Reply-To: <49D68DCB-057C-4E99-8E69-7FE730DC7F55@mail.grenander.com> Message-ID: <105534.56893.qm@web51610.mail.re2.yahoo.com> --- Keelan Lightfoot wrote: > What about the possibility of using a computer's > sound card as a 4 > wire trunk? FWIW, a long time ago now, I wrote an asterisk driver (an extended version of the then-existing chan_oss driver) that enabled the PC sound card to support two FXS ports (one each using the left and right channels). See http://www.lightlink.com/mhp/t/t.html Best Max p.s. I agree fully with the sentiment expressed - especially I would like to see increased use of SF pulsing within CNET ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs From stfkerman at jps.net Thu Dec 6 09:16:58 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 10:16:58 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <105534.56893.qm@web51610.mail.re2.yahoo.com> References: <105534.56893.qm@web51610.mail.re2.yahoo.com> Message-ID: <4758126A.60005@jps.net> I've noticed a surprising amount of crosstalk between the 2 channels on some PCs. I have not tested this methodically. Have you tried this and found the crosstalk to be low enough to really use the 2 channels independently? Considering the variability in dropouts I've experienced from one moment to the next and one call to the next on conversations I've had on CNET, I wonder how reliably SF would work. How much testing has been done? Steph ikjtel wrote: > FWIW, a long time ago now, I wrote an asterisk driver (an extended version of the then-existing chan_oss driver) that enabled the PC sound card to support two FXS ports (one each using the left and right channels). > > See http://www.lightlink.com/mhp/t/t.html > > Best > > Max > > p.s. I agree fully with the sentiment expressed - especially I would like to see increased use of SF pulsing within CNET > > --- Keelan Lightfoot > wrote: > > >> What about the possibility of using a computer's sound card as a 4 wire trunk? >> > > > > > From ikj1234i at yahoo.com Thu Dec 6 09:19:00 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Thu, 6 Dec 2007 07:19:00 -0800 (PST) Subject: [VoIP] Channel Bank on Ebay [not mine] and E&M In-Reply-To: <105534.56893.qm@web51610.mail.re2.yahoo.com> Message-ID: <933960.9730.qm@web51603.mail.re2.yahoo.com> Currently on Ebay there's a Coastcom channel bank with a total of 2 T1's and 48 FXS ports. Initial asking price is $19.99. [Item number: 120193396868] I have no interest in this item except that I have an identical Coastcom unit here which I can say is definitely Telco quality, *not* "consumer" grade! Manuals are freely downloadable in PDF format from coastcom.com. Additionally I have several 4W E&M cards and I would be very willing to consider sending a couple of them (for shipping costs only) to someone on list if they will use them... Each plugin card supports two separate 4W E&M trunks... Max ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping From madmanmarkau at hotmail.com Thu Dec 6 09:31:59 2007 From: madmanmarkau at hotmail.com (Mad Mark) Date: Thu, 6 Dec 2007 15:31:59 +0000 Subject: [VoIP] Channel Bank on Ebay [not mine] and E&M In-Reply-To: <933960.9730.qm@web51603.mail.re2.yahoo.com> References: <105534.56893.qm@web51610.mail.re2.yahoo.com> <933960.9730.qm@web51603.mail.re2.yahoo.com> Message-ID: I don't see any RJ9 (telephone) ports on it anywhere. How are phones hooked to such a unit?? > Date: Thu, 6 Dec 2007 07:19:00 -0800 > From: ikj1234i at yahoo.com > To: voip at ckts.info > Subject: [VoIP] Channel Bank on Ebay [not mine] and E&M > > > Currently on Ebay there's a Coastcom channel bank with > a total of 2 T1's and 48 FXS ports. Initial asking > price is $19.99. [Item number: 120193396868] > > I have no interest in this item except that I have an > identical Coastcom unit here which I can say is > definitely Telco quality, *not* "consumer" grade! > Manuals are freely downloadable in PDF format from > coastcom.com. > > Additionally I have several 4W E&M cards and I would > be very willing to consider sending a couple of them > (for shipping costs only) to someone on list if they > will use them... Each plugin card supports two > separate 4W E&M trunks... > > Max > > > ____________________________________________________________________________________ > Looking for last minute shopping deals? > Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ _________________________________________________________________ What are you waiting for? Join Lavalife FREE http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Flavalife9%2Eninemsn%2Ecom%2Eau%2Fclickthru%2Fclickthru%2Eact%3Fid%3Dninemsn%26context%3Dan99%26locale%3Den%5FAU%26a%3D30288&_t=764581033&_r=email_taglines_Join_free_OCT07&_m=EXT From ikj1234i at yahoo.com Thu Dec 6 09:32:28 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Thu, 6 Dec 2007 07:32:28 -0800 (PST) Subject: [VoIP] More "Authentic" CNET In-Reply-To: <4758126A.60005@jps.net> Message-ID: <516133.87580.qm@web51606.mail.re2.yahoo.com> --- Steph Kerman wrote: > I've noticed a surprising amount of crosstalk > between the 2 channels on > some PCs. I have not tested this methodically. > Have you tried this > and found the crosstalk to be low enough to really > use the 2 channels > independently? Speech is inaudible, DTMF tones in the opposite channel ARE barely audible. This could also have been caused by the substandard external circuit designs that I implemented... > > Considering the variability in dropouts I've > experienced from one moment > to the next and one call to the next on > conversations I've had on CNET, > I wonder how reliably SF would work. How much > testing has been done? My SF patch contains a low-pass filter that, in theory, can accomodate drop-outs (due to packet loss, say) of up to 15 msec. This should handle many cases of packet loss without hiccup. I did some testing into JN's PBX but my 2600 generator was not working properly, I think. Nonetheless I was able to dial shorter digits very sucessfully, and longer ones OK sometimes, and not OK sometimes. At that time the FXO connection was out of John's T1 channel bank, which was working well enough to forward audio *throughout* a dialled digit. Overall the experience was very similar to my memory of doing the same thing from a No 5. xbar into a CDO that was served over SF trunks (in area code 413 in 1978). The only thing missing was in the real network of old, there used to be a very slight 2600 tone audible in both directions on the trunk, as well as a slight crackle and hiss in the connection. Both were missed. I believe Jayson has tested doing 2600/SF outpulsing (successfully) also. > > Steph > > ikjtel wrote: > > FWIW, a long time ago now, I wrote an asterisk > driver (an extended version of the then-existing > chan_oss driver) that enabled the PC sound card to > support two FXS ports (one each using the left and > right channels). > > > > See http://www.lightlink.com/mhp/t/t.html > > > > Best > > > > Max > > > > p.s. I agree fully with the sentiment expressed - > especially I would like to see increased use of SF > pulsing within CNET > > > > --- Keelan Lightfoot > > wrote: > > > > > >> What about the possibility of using a computer's > sound card as a 4 wire trunk? > >> > > > > > > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ From ikj1234i at yahoo.com Thu Dec 6 09:37:02 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Thu, 6 Dec 2007 07:37:02 -0800 (PST) Subject: [VoIP] Channel Bank on Ebay [not mine] and E&M In-Reply-To: Message-ID: <463898.21777.qm@web51603.mail.re2.yahoo.com> Yeah, on the unit that I have, there are no "RJ" ports. Instead I have a 25-pair cable that's punched down onto a 66-type block. The cable plugs into the back of the channel bank using the same standard type of connector that are used on ordinary 1A2 key phone sets... Max --- Mad Mark wrote: > I don't see any RJ9 (telephone) ports on it > anywhere. How are phones hooked to such a unit?? > > > > > Date: Thu, 6 Dec 2007 07:19:00 -0800 > > From: ikj1234i at yahoo.com > > To: voip at ckts.info > > Subject: [VoIP] Channel Bank on Ebay [not mine] > and E&M > > > > > > Currently on Ebay there's a Coastcom channel bank > with > > a total of 2 T1's and 48 FXS ports. Initial > asking > > price is $19.99. [Item number: 120193396868] > > > > I have no interest in this item except that I have > an > > identical Coastcom unit here which I can say is > > definitely Telco quality, *not* "consumer" grade! > > Manuals are freely downloadable in PDF format from > > coastcom.com. > > > > Additionally I have several 4W E&M cards and I > would > > be very willing to consider sending a couple of > them > > (for shipping costs only) to someone on list if > they > > will use them... Each plugin card supports two > > separate 4W E&M trunks... > > > > Max > > > > > > > ____________________________________________________________________________________ > > Looking for last minute shopping deals? > > Find them fast with Yahoo! Search. > http://tools.search.yahoo.com/newsearch/category.php?category=shopping > > _______________________________________________ > > VoIP mailing list > > VoIP at ckts.info > > http://lists.ckts.info/mailman/listinfo/voip > > Project Web Page: http://www.ckts.info/ > > _________________________________________________________________ > What are you waiting for? Join Lavalife FREE > http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Flavalife9%2Eninemsn%2Ecom%2Eau%2Fclickthru%2Fclickthru%2Eact%3Fid%3Dninemsn%26context%3Dan99%26locale%3Den%5FAU%26a%3D30288&_t=764581033&_r=email_taglines_Join_free_OCT07&_m=EXT > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs From WATSON061502 at AMERITECH.NET Thu Dec 6 09:39:22 2007 From: WATSON061502 at AMERITECH.NET (Nathan Watson) Date: Thu, 06 Dec 2007 07:39:22 -0800 Subject: [VoIP] Channel Bank on Ebay [not mine] and E&M Message-ID: This isn't a consumer "plug n' play" item. You need to wire it up. Nathan --- Original Message --- From: Mad Mark To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Channel Bank on Ebay [not mine] and E&M >I don't see any RJ9 (telephone) ports on it anywhere. How are phones hooked to such a unit?? > > > >> Date: Thu, 6 Dec 2007 07:19:00 -0800 >> From: ikj1234i at yahoo.com >> To: voip at ckts.info >> Subject: [VoIP] Channel Bank on Ebay [not mine] and E&M >> >> >> Currently on Ebay there's a Coastcom channel bank with >> a total of 2 T1's and 48 FXS ports. Initial asking >> price is $19.99. [Item number: 120193396868] >> >> I have no interest in this item except that I have an >> identical Coastcom unit here which I can say is >> definitely Telco quality, *not* "consumer" grade! >> Manuals are freely downloadable in PDF format from >> coastcom.com. >> >> Additionally I have several 4W E&M cards and I would >> be very willing to consider sending a couple of them >> (for shipping costs only) to someone on list if they >> will use them... Each plugin card supports two >> separate 4W E&M trunks... >> >> Max From ikj1234i at yahoo.com Thu Dec 6 09:49:36 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Thu, 6 Dec 2007 07:49:36 -0800 (PST) Subject: [VoIP] More "Authentic" CNET In-Reply-To: <516133.87580.qm@web51606.mail.re2.yahoo.com> Message-ID: <686362.40744.qm@web51604.mail.re2.yahoo.com> --- ikjtel wrote: > [snip] was > able to dial shorter digits very sucessfully, and > longer ones OK sometimes, and not OK sometimes. I was convinced at the time that the cause of the "bad" digits had nothing whatsoever to do with audio drop-outs or VOIP packet loss. At that time John and others (Lee?, Kirt?) were having problems which were well-discussed on the list (channel bank FXO erratic outpulsing errors). The problem was audible as you could actually hear the pulse train sounded incorrect. I believe the root cause of the problem may have been in the Varion T1 product somewhere... Max ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping From voiptandem at shaneyoung.com Thu Dec 6 10:25:56 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Thu, 06 Dec 2007 10:25:56 -0600 Subject: [VoIP] Time and temp, weather, etc. In-Reply-To: <002501c8380c$d9d09e30$6900a8c0@BOE> References: <002501c8380c$d9d09e30$6900a8c0@BOE> Message-ID: <20071206102556.pylslzqogcc4goog@secure.shaneyoung.com> +1-821-7353 gives the weather for Minneapolis I have been working (slowly) on a time and temp machine based on Asterisk. +1-821-7301 gives the time and temp at my house in Minneapolis. Quoting Jayson Smith : > Hi, > > I think it would be cool if some people, myself included, could provide > real-time time and temperature, and possibly weather, service, without > having to place a call to a local number, since this call might cost money > if placed through a voip provider. Has anyone ever thought of designing > something like a thin client running Asterisk, Astlinux or whatever, with a > time and temperature app? How would the thin client receive the temperature > data? I actually do have a talking weather station with a wireless probe for > receiving temperature and relative humidity readings. Unfortunately, that's > about all that talks, other than a very short twelve to twenty-four hour > weather forecast based on barometric pressure. And as far as I know, there > are no audio outputs or anything, so it's just a stand-alone device. > Shane, I know you once offered to build a server with a time and temp > application for anyone who could supply recordings. How are you receiving > temperature data for your own T&T number? Has anyone ever recorded Alan's or > anyone else's T&T machine with Jane Barbe or similar announcements? Just a > few thoughts. > Jayson > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > --Shane +1-821-7311 CNET ---------------------------------------------------------------- From stfkerman at jps.net Thu Dec 6 10:33:56 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 11:33:56 -0500 Subject: [VoIP] Channel Bank on Ebay [not mine] and E&M In-Reply-To: References: <105534.56893.qm@web51610.mail.re2.yahoo.com> <933960.9730.qm@web51603.mail.re2.yahoo.com> Message-ID: <47582474.4090408@jps.net> RJ9? My Registration Service Manual jumps from RJA3 to RJ11. What is an RJ9? Steph Mad Mark wrote: > I don't see any RJ9 (telephone) ports on it anywhere. How are phones hooked to such a unit?? > > > > >> Date: Thu, 6 Dec 2007 07:19:00 -0800 >> From: ikj1234i at yahoo.com >> To: voip at ckts.info >> Subject: [VoIP] Channel Bank on Ebay [not mine] and E&M >> >> >> Currently on Ebay there's a Coastcom channel bank with >> a total of 2 T1's and 48 FXS ports. Initial asking >> price is $19.99. [Item number: 120193396868] >> >> I have no interest in this item except that I have an >> identical Coastcom unit here which I can say is >> definitely Telco quality, *not* "consumer" grade! >> Manuals are freely downloadable in PDF format from >> coastcom.com. >> >> Additionally I have several 4W E&M cards and I would >> be very willing to consider sending a couple of them >> (for shipping costs only) to someone on list if they >> will use them... Each plugin card supports two >> separate 4W E&M trunks... >> >> Max > From stfkerman at jps.net Thu Dec 6 10:45:57 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 11:45:57 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <686362.40744.qm@web51604.mail.re2.yahoo.com> References: <686362.40744.qm@web51604.mail.re2.yahoo.com> Message-ID: <47582745.5040806@jps.net> I recall that thread. I don't recall seeing anything about a cause having been determined. As I remember it, the pulses were unevenly timed, with pauses inserted into the pulse trains. I don't see how the channel bank could do anything other than shift the make/break ratio, which was not how I understood the problem having been described, but I never heard the sound of the aberrant outpulsing. Perhaps compelled MF would work better than SF because the compelled handshake would ensure correct digit transmission regardless of dropouts. Steph ikjtel wrote: > --- ikjtel wrote: >> [snip] was able to dial shorter digits very sucessfully, and longer >> ones OK sometimes, and not OK sometimes. > I was convinced at the time that the cause of the "bad" digits had > nothing whatsoever to do with audio drop-outs or VOIP packet loss. > > At that time John and others (Lee?, Kirt?) were having problems which > were well-discussed on the list (channel bank FXO erratic outpulsing > errors). The problem was audible as you could actually hear the pulse > train sounded incorrect. I believe the root cause of the problem may > have been in the Varion T1 product somewhere... > > Max From ikj1234i at yahoo.com Thu Dec 6 11:06:50 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Thu, 6 Dec 2007 09:06:50 -0800 (PST) Subject: [VoIP] More "Authentic" CNET In-Reply-To: <47582745.5040806@jps.net> Message-ID: <114351.37072.qm@web51606.mail.re2.yahoo.com> Yeah, I was able to make (what I believed was) a complete list of all variables in play including product/model of hardware as well as version(s) of software used by Lee, John, and Kirt. I then set up a testbed here matching every piece (except I used a genuine Digium T1 card and I used a different brand of channel bank) and verified that all software versions were the same. Inasmuch as I was unable to reproduce the problem here, the conclusion was that all variables could therefore be eliminated except two: the channel bank brand/model, and the varion T1 card. I then made a sort of informal (but educated) guess that if I were betting I would choose to wager that, of the two, the varion card was the culprit. Also, additional ammunition for this is the fact that varion has some sketchy "modifications" in that very area, having to do with "a" and "b" timing bits or some such... In any case, I don't feel that one isolated case of bad hardware/software should dictate what kinds of cool features we should or should not add to CNET........ Max --- Steph Kerman wrote: > I recall that thread. I don't recall seeing > anything about a cause > having been determined. As I remember it, the > pulses were unevenly > timed, with pauses inserted into the pulse trains. > I don't see how the > channel bank could do anything other than shift the > make/break ratio, > which was not how I understood the problem having > been described, but I > never heard the sound of the aberrant outpulsing. > > Perhaps compelled MF would work better than SF > because the compelled > handshake would ensure correct digit transmission > regardless of dropouts. > > Steph > > ikjtel wrote: > > --- ikjtel wrote: > >> [snip] was able to dial shorter digits very > sucessfully, and longer > >> ones OK sometimes, and not OK sometimes. > > I was convinced at the time that the cause of the > "bad" digits had > > nothing whatsoever to do with audio drop-outs or > VOIP packet loss. > > > > At that time John and others (Lee?, Kirt?) were > having problems which > > were well-discussed on the list (channel bank FXO > erratic outpulsing > > errors). The problem was audible as you could > actually hear the pulse > > train sounded incorrect. I believe the root cause > of the problem may > > have been in the Varion T1 product somewhere... > > > > Max > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping From ratguy at insightbb.com Thu Dec 6 11:11:05 2007 From: ratguy at insightbb.com (Jayson Smith) Date: Thu, 6 Dec 2007 12:11:05 -0500 Subject: [VoIP] Time and temp, weather, etc. References: <002501c8380c$d9d09e30$6900a8c0@BOE> <20071206102556.pylslzqogcc4goog@secure.shaneyoung.com> Message-ID: <000801c8382a$fcbdfaf0$6900a8c0@BOE> 821-7301 is giving me an enum failure at my end.... Jayson ----- Original Message ----- From: "Shane Young" To: Sent: Thursday, December 06, 2007 11:25 AM Subject: Re: [VoIP] Time and temp, weather, etc. > +1-821-7353 gives the weather for Minneapolis > > I have been working (slowly) on a time and temp machine based on Asterisk. > > +1-821-7301 gives the time and temp at my house in Minneapolis. > > Quoting Jayson Smith : > >> Hi, >> >> I think it would be cool if some people, myself included, could >> provide >> real-time time and temperature, and possibly weather, service, without >> having to place a call to a local number, since this call might cost >> money >> if placed through a voip provider. Has anyone ever thought of designing >> something like a thin client running Asterisk, Astlinux or whatever, with >> a >> time and temperature app? How would the thin client receive the >> temperature >> data? I actually do have a talking weather station with a wireless probe >> for >> receiving temperature and relative humidity readings. Unfortunately, >> that's >> about all that talks, other than a very short twelve to twenty-four hour >> weather forecast based on barometric pressure. And as far as I know, >> there >> are no audio outputs or anything, so it's just a stand-alone device. >> Shane, I know you once offered to build a server with a time and >> temp >> application for anyone who could supply recordings. How are you receiving >> temperature data for your own T&T number? Has anyone ever recorded Alan's >> or >> anyone else's T&T machine with Jane Barbe or similar announcements? Just >> a >> few thoughts. >> Jayson >> _______________________________________________ >> VoIP mailing list >> VoIP at ckts.info >> http://lists.ckts.info/mailman/listinfo/voip >> Project Web Page: http://www.ckts.info/ >> > > --Shane > +1-821-7311 CNET > > ---------------------------------------------------------------- > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > From stfkerman at jps.net Thu Dec 6 11:23:29 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 12:23:29 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <114351.37072.qm@web51606.mail.re2.yahoo.com> References: <114351.37072.qm@web51606.mail.re2.yahoo.com> Message-ID: <47583011.4080602@jps.net> Certainly, the negative proof coming out of *not* being able to reproduce the problem after replacing all elements except certain specific ones has a good degree of uncertainty, but undoubtedly you know that. Now that I understand more clearly that the Varion is a card, presumably with it's own set of drivers, I agree that there is a greater chance that it is an element or the culprit in the problem that was observed. I previously understood that the name referred to a channel bank. By "sketchy modification" are you referring to driver code modifications or hardware modifications? I agree that isolated trouble cases should not be the basis for restricting the scope of what should be attempted lacking information that these cases reveal some inherent limitation. I'd think they should be a cautionary sign to investigate further. Steph ikjtel wrote: > Yeah, I was able to make (what I believed was) a complete list of all > variables in play including product/model of hardware as well as > version(s) of software used by Lee, John, and Kirt. > > I then set up a testbed here matching every piece (except I used a > genuine Digium T1 card and I used a different brand of channel bank) > and verified that all software versions were the same. Inasmuch as I > was unable to reproduce the problem here, the conclusion was that all > variables could therefore be eliminated except two: the channel bank > brand/model, and the varion T1 card. I then made a sort of informal > (but educated) guess that if I were betting I would choose to wager > that, of the two, the varion card was the culprit. Also, additional > ammunition for this is the fact that varion has some sketchy > "modifications" in that very area, having to do with "a" and "b" > timing bits or some such... > > In any case, I don't feel that one isolated case of bad > hardware/software should dictate what kinds of cool features we should > or should not add to CNET........ > > Max > > Steph Kerman wrote: >> I recall that thread. I don't recall seeing anything about a cause >> having been determined. As I remember it, the pulses were unevenly >> timed, with pauses inserted into the pulse trains. I don't see how >> the channel bank could do anything other than shift the make/break >> ratio, which was not how I understood the problem having been >> described, but I never heard the sound of the aberrant outpulsing. >> >> Perhaps compelled MF would work better than SF because the compelled >> handshake would ensure correct digit transmission regardless of dropouts. >> >> Steph >> >> >> ikjtel wrote: >>> --- ikjtel wrote: >>> >>> >>>> [snip] was >>>> able to dial shorter digits very sucessfully, and longer ones OK >>>> sometimes, and not OK sometimes. >>> >>> I was convinced at the time that the cause of the "bad" digits had >>> nothing whatsoever to do with audio drop-outs or VOIP packet loss. >>> >>> At that time John and others (Lee?, Kirt?) were having problems >>> which were well-discussed on the list (channel bank FXO erratic >>> outpulsing errors). The problem was audible as you could actually >>> hear the pulse train sounded incorrect. I believe the root cause of >>> the problem may have been in the Varion T1 product somewhere... >>> >>> Max >>> From stfkerman at jps.net Thu Dec 6 11:29:33 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 12:29:33 -0500 Subject: [VoIP] More "Authentic" CNET (cont'd) In-Reply-To: <114351.37072.qm@web51606.mail.re2.yahoo.com> References: <114351.37072.qm@web51606.mail.re2.yahoo.com> Message-ID: <4758317D.5010408@jps.net> ..... I agree that isolated trouble cases should not be the basis for restricting the scope of what should be attempted lacking information that these cases reveal some inherent limitation. I'd think they should be a cautionary sign to investigate further. And you seem to have pursued it to the extent reasonably practical under the circumstances. Steph From stfkerman at jps.net Thu Dec 6 11:41:31 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 12:41:31 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <105534.56893.qm@web51610.mail.re2.yahoo.com> References: <105534.56893.qm@web51610.mail.re2.yahoo.com> Message-ID: <4758344B.4070808@jps.net> Here's something that seems relevant to this thread: http://www.herbach.com/Merchant2/merchant.mv?Screen=PROD&Store_Code=HAR&Product_Code=RB-213&Category_Code=KIT I wonder what it consists of. Steph ikjtel wrote: > --- Keelan Lightfoot > wrote: > > >> What about the possibility of using a computer's sound card as a 4 wire trunk? >> > > > FWIW, a long time ago now, I wrote an asterisk driver (an extended > version of the then-existing chan_oss driver) that enabled the PC > sound card to support two FXS ports (one each using the left and right > channels). > > See http://www.lightlink.com/mhp/t/t.html > > Best > > Max > > p.s. I agree fully with the sentiment expressed - especially I would > like to see increased use of SF pulsing within CNET > From ikj1234i at yahoo.com Thu Dec 6 11:58:09 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Thu, 6 Dec 2007 09:58:09 -0800 (PST) Subject: [VoIP] More "Authentic" CNET In-Reply-To: <47583011.4080602@jps.net> Message-ID: <886262.29871.qm@web51609.mail.re2.yahoo.com> --- Steph Kerman wrote: > Certainly, the negative proof coming out of *not* > being able to > reproduce the problem after replacing all elements > except certain > specific ones has a good degree of uncertainty, but > undoubtedly you know > that. Correct. Ideally a more scientific test would have started out with an identical system that was having the same problem. Then, you change variables exactly one at a time until the problem changes / goes away. In this case I didn't have the needed hardware to do that. However the problem where it occured was happening on nearly every digit dialled so should have been very easy to reproduce if it was going to happen. > > Now that I understand more clearly that the Varion > is a card, presumably > with it's own set of drivers, I agree that there is > a greater chance > that it is an element or the culprit in the problem > that was observed. > I previously understood that the name referred to a > channel bank. > > By "sketchy modification" are you referring to > driver code modifications > or hardware modifications? Software. I can't speak for the hardware, perhaps others may know more. However as to the software, you have to apply a varion vendor-supplied driver patch and/or other software driver modifications on top of the standard zaptel code base. Max ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs From lee at spenadel.com Thu Dec 6 12:00:28 2007 From: lee at spenadel.com (Lee Spenadel) Date: Thu, 6 Dec 2007 13:00:28 -0500 Subject: [VoIP] Weather by Zip Message-ID: <006d01c83831$e35e2510$aa1a6f30$@com> I seemed to have fallen off the list serve, but John Novack told me there was some discussion on whether I have a weather application. I do. Dial 349-1616 and enter your zip code. Lee From stfkerman at jps.net Thu Dec 6 12:06:12 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 13:06:12 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <886262.29871.qm@web51609.mail.re2.yahoo.com> References: <886262.29871.qm@web51609.mail.re2.yahoo.com> Message-ID: <47583A14.3050203@jps.net> ikjtel wrote: > --- Steph Kerman wrote: >> Certainly, the negative proof coming out of *not* being able to >> reproduce the problem after replacing all elements except certain >> specific ones has a good degree of uncertainty, but undoubtedly you >> know that. > Correct. Ideally a more scientific test would have started out with an > identical system that was having the same problem. Then, you change > variables exactly one at a time until the problem changes / goes away. > > In this case I didn't have the needed hardware to do that. Understood. One interesting question is whether anyone else using the same hardware did *not* have the problem. > However the problem where it occured was happening on nearly every > digit dialled so should have been very easy to reproduce if it was > going to happen. Do you have any WAV files that could be used to hear what it sounded like? >> Now that I understand more clearly that the Varion is a card, >> presumably with it's own set of drivers, I agree that there is a >> greater chance that it is an element or the culprit in the problem >> that was observed. I previously understood that the name referred to a >> channel bank. >> >> By "sketchy modification" are you referring to driver code >> modifications or hardware modifications? > Software. I can't speak for the hardware, perhaps others may know > more. However as to the software, you have to apply a varion > vendor-supplied driver patch and/or other software driver > modifications on top of the standard zaptel code base. Ugh! Steph From ratguy at insightbb.com Thu Dec 6 12:09:25 2007 From: ratguy at insightbb.com (Jayson Smith) Date: Thu, 6 Dec 2007 13:09:25 -0500 Subject: [VoIP] Weather by Zip References: <006d01c83831$e35e2510$aa1a6f30$@com> Message-ID: <000b01c83833$225fcb50$6900a8c0@BOE> I know about that. However, it has a few disadvantages. First, Festival isn't, imho, the best speech that could be used, although it's probably one of few free options, if not the only one. That is, unless you happen to have an old DECtalk DTC-01 or similar sitting around. Also, what I'm really looking for is a way where anybody could call a number in *my* office code and get *my* local forecast, without me having to forward their call to the local weather number. Anyone know of an IAX/SIP-based weather service where you, say, pass zip code as name in CID and it provides an audio weather report? Jayson ----- Original Message ----- From: "Lee Spenadel" To: "'Voice Over IP Tandem for Analog Switches'" Sent: Thursday, December 06, 2007 1:00 PM Subject: [VoIP] Weather by Zip >I seemed to have fallen off the list serve, but John Novack told me there > was some discussion on whether I have a weather application. I do. Dial > 349-1616 and enter your zip code. > > > > Lee > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > From jnovack at stromberg-carlson.org Thu Dec 6 09:17:25 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Thu, 06 Dec 2007 10:17:25 -0500 Subject: [VoIP] Time and temp, weather, etc. In-Reply-To: <002501c8380c$d9d09e30$6900a8c0@BOE> References: <002501c8380c$d9d09e30$6900a8c0@BOE> Message-ID: <47581285.60304@stromberg-carlson.org> Jayson Smith wrote: > Hi, > > I think it would be cool if some people, myself included, could provide real-time time and temperature, and possibly weather, service, without having to place a call to a local number, since this call might cost money if placed through a voip provider. Has anyone ever thought of designing something like a thin client running Asterisk, Astlinux or whatever, with a time and temperature app? Lee Spenadel has, or had, done just this on his box. I don't know if it is currently active, but his asked for the zip code and went to the NWS for the current weather. Quite a task to set up, as I remember, hazy now, but it worked. Would have to revisit to see if it could work with AstLinux, which has some limitations, but is quite amazing in what DOES work in a less than 64 Meg CF card. John Novack -- Dog is my co-pilot From ikj1234i at yahoo.com Thu Dec 6 12:20:05 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Thu, 6 Dec 2007 10:20:05 -0800 (PST) Subject: [VoIP] More "Authentic" CNET In-Reply-To: <47583A14.3050203@jps.net> Message-ID: <21403.80559.qm@web51608.mail.re2.yahoo.com> --- Steph Kerman wrote: > Do you have any WAV files that could be used to hear > what it sounded like? I don't. Not sure whether any of the folks who were having this problem still have any of the equipment hooked up in a way that you could hear the problem. However it's definitely easily perceptible. [Note. I ingested certain substances at University - the statute of limitations has long since expired - that enabled me to easily perceive the 60 Hz flicker of fluorescent lights, accelerating the neurons so the 60 Hz pulsations seemed almost slow. By comparison, 10 Hz dial pulsing has been glacially slow ever since. -YMMV] Also, it should be noted that there were patches floating around to change the default zaptel outpulsing ratios from (standard but incorrect zaptel values of) 50/50 to a more correct value of 60/40. Some users reported improvement, but not complete success, if I recall correctly. In my view the 50/50 vs. 60/40 changes were a complete red herring with respect to the classic "erratic FXO outpulsing" problem... Max ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping From voiptandem at shaneyoung.com Thu Dec 6 12:21:38 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Thu, 06 Dec 2007 12:21:38 -0600 Subject: [VoIP] Weather by Zip In-Reply-To: <000b01c83833$225fcb50$6900a8c0@BOE> References: <006d01c83831$e35e2510$aa1a6f30$@com> <000b01c83833$225fcb50$6900a8c0@BOE> Message-ID: <20071206122138.42dy50rc8oo4cg0o@secure.shaneyoung.com> The same application could be put onto everyone's system and have the zip hard-coded so it doesn't need to be entered. Quoting Jayson Smith : > I know about that. However, it has a few disadvantages. First, Festival > isn't, imho, the best speech that could be used, although it's probably one > of few free options, if not the only one. That is, unless you happen to have > an old DECtalk DTC-01 or similar sitting around. Also, what I'm really > looking for is a way where anybody could call a number in *my* office code > and get *my* local forecast, without me having to forward their call to the > local weather number. Anyone know of an IAX/SIP-based weather service where > you, say, pass zip code as name in CID and it provides an audio weather > report? > Jayson > > ----- Original Message ----- > From: "Lee Spenadel" > To: "'Voice Over IP Tandem for Analog Switches'" > Sent: Thursday, December 06, 2007 1:00 PM > Subject: [VoIP] Weather by Zip > > >> I seemed to have fallen off the list serve, but John Novack told me there >> was some discussion on whether I have a weather application. I do. Dial >> 349-1616 and enter your zip code. >> >> >> >> Lee >> >> >> >> >> >> _______________________________________________ >> VoIP mailing list >> VoIP at ckts.info >> http://lists.ckts.info/mailman/listinfo/voip >> Project Web Page: http://www.ckts.info/ >> > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > --Shane +1-821-7311 CNET ---------------------------------------------------------------- From stfkerman at jps.net Thu Dec 6 12:31:33 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 06 Dec 2007 13:31:33 -0500 Subject: [VoIP] More "Authentic" CNET In-Reply-To: <21403.80559.qm@web51608.mail.re2.yahoo.com> References: <21403.80559.qm@web51608.mail.re2.yahoo.com> Message-ID: <47584005.1090907@jps.net> ikjtel wrote: > --- Steph Kerman wrote: >> Do you have any WAV files that could be used to hear what it sounded >> like? > I don't. Not sure whether any of the folks who were having this > problem still have any of the equipment hooked up in a way that you > could hear the problem. However it's definitely easily perceptible. > > [Note. I ingested certain substances at University - the statute of > limitations has long since expired - that enabled me to easily > perceive the 60 Hz flicker of fluorescent lights, accelerating the > neurons so the 60 Hz pulsations seemed almost slow. By comparison, 10 > Hz dial pulsing has been glacially slow ever since. -YMMV] I have never been convinced that those effects were anything other than subjective and ephemeral, but.... "YMMV". That said, I used to adjust the A relays on my connector switches for correct pulsing and B relay residual screw for correct release time by ear. I was in my mid/late teens at the time and as of that time had never consumed any of the substances you are referring to. > Also, it should be noted that there were patches floating around to > change the default zaptel outpulsing ratios from (standard but > incorrect zaptel values of) 50/50 to a more correct value of 60/40. > Some users reported improvement, but not complete success, if I recall > correctly. In my view the 50/50 vs. 60/40 changes were a complete red > herring with respect to the classic "erratic FXO outpulsing" problem... As with any m