From jnovack at stromberg-carlson.org Mon Jan 1 11:04:10 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Mon, 01 Jan 2007 12:04:10 -0500 Subject: [VoIP] Last new CNET membber of 2006 Message-ID: <45993F0A.1060500@stromberg-carlson.org> Lee Spenadel, office code 349, AFAIK was the last new member to join CNET in 2006 Last afternoon/evening we got his FXS ATA working over CNET Lee and I will be working on more items today and perhaps even getting is homebrew SXS switch connected and working, as well as his Merlin system Looking forward to more growth of CNET in 2007 Hope you all have a Happy New Year John Novack From axg at syntec.co.uk Tue Jan 2 15:25:46 2007 From: axg at syntec.co.uk (Andy) Date: Tue, 2 Jan 2007 21:25:46 -0000 Subject: [VoIP] Sipgate Message-ID: <008a01c72eb4$912c0280$8658b281@andy> Gents, I have a sipgate number, and interestingly ever since it has been on I have had problems. Yesterday my adsl router locked up and needed a reboot, and today I've had to reboot both the router and the asterisk box. There must be something to do with their pinging that is causing this as for almost a year my box has been up with no problems. Can anyone shed any light on this? Andy G -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.16.2/613 - Release Date: 01/01/2007 14:50 From chad at maine.maine.edu Tue Jan 2 21:31:16 2007 From: chad at maine.maine.edu (Chad Perkins) Date: Tue, 02 Jan 2007 22:31:16 -0500 Subject: [VoIP] Last new CNET membber of 2006 In-Reply-To: <45993F0A.1060500@stromberg-carlson.org> Message-ID: <459ADD34.3107.68642A@localhost> We need to work directory page(s) folks. We continue to get more folks on CNET but if you're not listed no one call you and say hi or listen to your machine(s). '363' and '377' hint, hint. Chad On 1 Jan 2007 at 12:04, John Novack wrote: > Lee Spenadel, office code 349, AFAIK was the last new member to join > CNET in 2006 Last afternoon/evening we got his FXS ATA working over > CNET Lee and I will be working on more items today and perhaps even > getting is homebrew SXS switch connected and working, as well as his > Merlin system > > Looking forward to more growth of CNET in 2007 > Hope you all have a Happy New Year > John Novack From stfkerman at jps.net Tue Jan 2 21:37:43 2007 From: stfkerman at jps.net (Steph Kerman) Date: Tue, 02 Jan 2007 22:37:43 -0500 Subject: [VoIP] DP-compatible ATAs In-Reply-To: <4596F9E5.21079.29BE7DB@localhost> References: <45954C42.6563.599D269@localhost> <4596F9E5.21079.29BE7DB@localhost> Message-ID: <459B2507.7030203@jps.net> Chad Perkins wrote: > On 29 Dec 2006 at 21:01, Steph Kerman wrote: >> Asterisk DP outpulsing will not produce authentic sounding DP towards >> the caller as was generally heard for DID completions to SXS >> Centrex-CUs. While it may not mask it, if the wirespring DP sender >> has to be there anyway to produce authentic sounds, the * becomes >> dispensible for this purpose. And the W-S senders are infinitely more >> fun to watch! > My assertion is that Asterisk (natively) will behave the same as any > modern PBX. If I have a Definity or a Nortel I'm not going to hear > anything different as the call is routed through the switch. I do > understand, and agree that if everything is on analog 2W copper and > "historic" sound is important, Asterisk does not contribute to that end. That depends. What is the propagation delay through an * switch? Steph From jjones3601 at yahoo.com Tue Jan 2 22:31:25 2007 From: jjones3601 at yahoo.com (john jones) Date: Tue, 2 Jan 2007 20:31:25 -0800 (PST) Subject: [VoIP] Update on Linksys Router running Asterisk Message-ID: <20070103043125.16226.qmail@web34302.mail.mud.yahoo.com> Good news. I have Asterisk running in a pretty stable mode (actually very stable AFAICT). I have interfaced it to a LinkSys PAP2-NA dual port FXS ATA, a Sipura 3000 with 1 FXS port and 1 FXO port, and a X-Lite softclient running on my laptop. Boy I fought that SPA3K and the working config sure seems like very close to what I started with! I don't think I can run two * boxes on the same (single) dynamic IP address so I'm going to take the existing 688 exchange off the air for a bit and bring up 687 on the Linksys. Eventually, I guess I'll dedicate the Linksys * box to the workshop when the EM swicth is and tandem through the other one. I also have a Cisco ATA that I'm trying to get unlocked and Steph is going to send me a MediaTrix ATA so I'll try to get both of them tested with the Linksys. I'm also going to buy another WRTSL54GS router because I believe that I can VERY easily make a clone of this system and then make minor edits to customize for someone else. I'll make a installation/customization document available along with this clone image if anyone is interested (probably a few weeks out still) John From stfkerman at jps.net Tue Jan 2 23:42:45 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 03 Jan 2007 00:42:45 -0500 Subject: [VoIP] Update on Linksys Router running Asterisk In-Reply-To: <20070103043125.16226.qmail@web34302.mail.mud.yahoo.com> References: <20070103043125.16226.qmail@web34302.mail.mud.yahoo.com> Message-ID: <459B4255.2080105@jps.net> I set up a simple network between 2 of my machines and one of the 2 port Mediatrix units so that I can talk to the Mediatrix unit. The unit has a configuration interface that you talk to with a browser. This allows country-specific configuration (call progress tone plan and ringing frequency) and other characteristics to be set. It seems that the Mediatrix APA allows Voip networks to be built using the APAs themselves and "APA Server software". They seem to be designed to function as a seamless extension of the PSTN. The APA FXS ports are programmed with dialing rules to determine the level of local and toll access to the PSTN.. The APA FXO ports can be programmed to function like a DISA, where incoming PSTN callers are answered with dial tone from the APA, whereupon they can dial any telephone on the Voip network. They can also be programmed to automatically extend an incoming PSTN call to any local FXS port on the same APA or any other APA on the Voip network. The manual that describes all this is in hard copy form. I do not have and have not found a PDF version of it on the Mediatrix website. Perhaps that is because they were rated "end of life" about 1 year ago and support was ended. I have one unit ready to send that I packed up for you last week but now that I am playing with another one I doubt you will get anywhere without a manual. I'm not certain that these units use standard protocols over the network to communicate with the APA server or each other. We might even need some kind of LAN protocol analyzer to understand what protocol they use. If so, they may require too much effort to integrate them with *. I'll let you know where this leads. Steph john jones wrote: > I also have a Cisco ATA that I'm trying to get unlocked and Steph is > going to send me a MediaTrix ATA so I'll try to get both of them > tested with the Linksys. > > From jjones3601 at yahoo.com Wed Jan 3 07:48:22 2007 From: jjones3601 at yahoo.com (john jones) Date: Wed, 3 Jan 2007 05:48:22 -0800 (PST) Subject: [VoIP] Dial Pulse converter specifications Message-ID: <20070103134822.60650.qmail@web34304.mail.mud.yahoo.com> Since so many VoIP ATA's don't support DP dialing, there has been a suggestion to construct a TT to DP converter. I'd like to start collecting specifications of what this device needs to do. I expect that on the FXS side, it ought to perform in ways similar to the teltone converters often installed between Linefinders and the 1st selector although I imagine that more loop supervision is required. Let's get a dialog going on this. Thanks! John From hockd at dteenergy.com Wed Jan 3 07:50:09 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Wed, 3 Jan 2007 08:50:09 -0500 Subject: [VoIP] Last new CNET membber of 2006 Message-ID: Chad, You are right we do need to pay attention to the directory. Hopefully we don't end up with the equivalent of the Moscow Directory huh. Ihave updated my listings sometime ago but seem to keep having trouble with the * box shgutting itself down. I need to speak to John about the configs. Take care to all Happy 2007, 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: 01/02/2007 10:31PM Subject: Re: [VoIP] Last new CNET membber of 2006 We need to work directory page(s) folks. We continue to get more folks on CNET but if you're not listed no one call you and say hi or listen to your machine(s). '363' and '377' hint, hint. Chad On 1 Jan 2007 at 12:04, John Novack wrote: > Lee Spenadel, office code 349, AFAIK was the last new member to join > CNET in 2006 Last afternoon/evening we got his FXS ATA working over > CNET Lee and I will be working on more items today and perhaps even > getting is homebrew SXS switch connected and working, as well as his > Merlin system > > Looking forward to more growth of CNET in 2007 > Hope you all have a Happy New Year > John Novack _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From stfkerman at jps.net Wed Jan 3 08:40:06 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 03 Jan 2007 09:40:06 -0500 Subject: [VoIP] Dial Pulse converter specifications In-Reply-To: <20070103134822.60650.qmail@web34304.mail.mud.yahoo.com> References: <20070103134822.60650.qmail@web34304.mail.mud.yahoo.com> Message-ID: <459BC046.8030307@jps.net> More loop supervision? What do you mean? 1) It needs to connect into the loop. A design was published that installed in a phone and connected directly to the dial pulse contacts. This is not a suitable configuration. 2) It needs to provide a DC current path during the dial pulse break intervals to prevent the downstream device from seeing each pulse as a disconnect. 3) It should support 10 PPS to 20 PPS operation. 20 PPS operation is not supported on some modern exchanges that support 10 PPS operation. Supporting 20 PPS operation would make it possible to use 20 PPS dials both in these environments and in SXS environments equipped with TT converters. 4) It should work with real-world dial pulses that have pulse characteristics at the extremes of the readjustment limits published by the Bell System. This includes ratio, average and instantaneous speed and contact bounce. 5) A goal would be to make it loop powered, but that's not an absolute necessity. It could be designed to only consume power during pulse conversion, so that battery operation could be achieved with very long battery life. 6) It should use readily available commercial electronic components: no antique switching equipment relays with pure unobtainium contacts. 7) A printed circuit board would be designed and procured from a commercial prototype board house to permit easy assembly without random hand wiring. Steph john jones wrote: > Since so many VoIP ATA's don't support DP dialing, > there has been a suggestion to construct a TT to DP > converter. > > I'd like to start collecting specifications of what > this device needs to do. > > I expect that on the FXS side, it ought to perform in > ways similar to the teltone converters often installed > between Linefinders and the 1st selector although I > imagine that more loop supervision is required. > > Let's get a dialog going on this. > > Thanks! > > John > From jnovack at stromberg-carlson.org Wed Jan 3 10:35:53 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Wed, 03 Jan 2007 11:35:53 -0500 Subject: [VoIP] Last new CNET membber of 2006 In-Reply-To: <459ADD34.3107.68642A@localhost> References: <459ADD34.3107.68642A@localhost> Message-ID: <459BDB69.9060101@stromberg-carlson.org> SOME of the folks are still doing some testing and making configuration changes. In Lee's case we just got an issue resolved, he needs to relocate his box and connect his switch. Patience is a virtue! John Novack Chad Perkins wrote: > We need to work directory page(s) folks. We continue to get more folks on CNET but if you're not listed no one call you and say hi or listen to your machine(s). > > '363' and '377' hint, hint. > > Chad > > On 1 Jan 2007 at 12:04, John Novack wrote: > >> Lee Spenadel, office code 349, AFAIK was the last new member to join >> CNET in 2006 Last afternoon/evening we got his FXS ATA working over >> CNET Lee and I will be working on more items today and perhaps even >> getting is homebrew SXS switch connected and working, as well as his >> Merlin system >> >> Looking forward to more growth of CNET in 2007 >> Hope you all have a Happy New Year >> John Novack >> > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > From erwin at windekind.demon.nl Wed Jan 3 10:59:13 2007 From: erwin at windekind.demon.nl (Erwin Dokter) Date: Wed, 3 Jan 2007 17:59:13 +0100 Subject: [VoIP] Configuring Asterisk Message-ID: <000d01c72f58$7f2cabd0$0501a8c0@windekind.demon.nl> Since Remco is still having problems with his * box, I decided to play with Asterisk myself a little. Since I don't have Linux, I installed AsteriskWin32, which is a port from Asterisk 1.0.10. I need a sample configuration to get it working for CNET. The setup is dead simple: ATA > Asterisk > CNET, no FSX cards and such. So I need a minimal configuration that excepts calls from CNET and allows me to call to CNET. I just activated my office code (31)318 via email. Any help is appriciated. -- Erwin Dokter From jnovack at stromberg-carlson.org Wed Jan 3 11:13:19 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Wed, 03 Jan 2007 12:13:19 -0500 Subject: [VoIP] Configuring Asterisk In-Reply-To: <000d01c72f58$7f2cabd0$0501a8c0@windekind.demon.nl> References: <000d01c72f58$7f2cabd0$0501a8c0@windekind.demon.nl> Message-ID: <459BE42F.70607@stromberg-carlson.org> Assumptions: You have a spare computer, it can boot from CD, and you can burn CD's from iso files You probably will have LOTS less problems with plain vanilla Linux Go to the CentOS site, download the ISO files for CentOS 3, ( I had problems with CentOS 4 and something called udev ) and burn the installation CD's Boot from CD # 1 go through the install and make sure you install everything. The Linux guru's will say that isn't necessary, but down the road when you want to install something and some little file or package is missing. When the machine is up and running with it's somewhat lame GUI, log on and go find Webmin with Mozilla/Firefox and follow the instructions to install. Again the Linux gurus on the list will be retching about now, but for the rest of us, Webmin gives access from your Windows machine for MANY configuration changes. Report back for lesson # 2 John Novack Erwin Dokter wrote: > Since Remco is still having problems with his * box, I decided to play with > Asterisk myself a little. Since I don't have Linux, I installed > AsteriskWin32, which is a port from Asterisk 1.0.10. I need a sample > configuration to get it working for CNET. > > The setup is dead simple: ATA > Asterisk > CNET, no FSX cards and such. So I > need a minimal configuration that excepts calls from CNET and allows me to > call to CNET. I just activated my office code (31)318 via email. > > Any help is appriciated. > From jnovack at stromberg-carlson.org Wed Jan 3 11:17:30 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Wed, 03 Jan 2007 12:17:30 -0500 Subject: [VoIP] TCI and switchers Quarterly Message-ID: <459BE52A.3000701@stromberg-carlson.org> Anyone on the list not yet renewed their subscriptions to TCI and Switchers Quarterly? Don't miss out ! Also, anyone want to contribute switching, technical or CNET related articles, the editor is ready and waiting for submissions Chris is technically challenged, so submissions to him with need to be in the form of fax or snail mail, but Doug Alderdice can receive via E-mail John Novack From jnovack at stromberg-carlson.org Wed Jan 3 11:20:48 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Wed, 03 Jan 2007 12:20:48 -0500 Subject: [VoIP] Configuring Asterisk In-Reply-To: <459BE42F.70607@stromberg-carlson.org> References: <000d01c72f58$7f2cabd0$0501a8c0@windekind.demon.nl> <459BE42F.70607@stromberg-carlson.org> Message-ID: <459BE5F0.2030903@stromberg-carlson.org> Regardless of what way you go, with no TDM cards, Asterisk needs a timing reference, and would have to have been compiled with ztdummy. The IAX protocol used with CNET requires that reference. How well this port works is questionable, given the addmittedly biased response es on the Asterisk users list. I can help you off list with configurations, if necessary John Novack wrote: > Assumptions: > You have a spare computer, it can boot from CD, and you can burn CD's > from iso files > You probably will have LOTS less problems with plain vanilla Linux > Go to the CentOS site, download the ISO files for CentOS 3, ( I had > problems with CentOS 4 and something called udev ) and burn the > installation CD's Boot from CD # 1 go through the install and make sure > you install everything. The Linux guru's will say that isn't necessary, > but down the road when you want to install something and some little > file or package is missing. > When the machine is up and running with it's somewhat lame GUI, log on > and go find Webmin with Mozilla/Firefox and follow the instructions to > install. Again the Linux gurus on the list will be retching about now, > but for the rest of us, Webmin gives access from your Windows machine > for MANY configuration changes. > Report back for lesson # 2 > > John Novack > > > Erwin Dokter wrote: > >> Since Remco is still having problems with his * box, I decided to play with >> Asterisk myself a little. Since I don't have Linux, I installed >> AsteriskWin32, which is a port from Asterisk 1.0.10. I need a sample >> configuration to get it working for CNET. >> >> The setup is dead simple: ATA > Asterisk > CNET, no FSX cards and such. So I >> need a minimal configuration that excepts calls from CNET and allows me to >> call to CNET. I just activated my office code (31)318 via email. >> >> Any help is appriciated. >> >> > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > From jjones3601 at yahoo.com Wed Jan 3 12:14:36 2007 From: jjones3601 at yahoo.com (john jones) Date: Wed, 3 Jan 2007 10:14:36 -0800 (PST) Subject: [VoIP] Configuring Asterisk In-Reply-To: <000d01c72f58$7f2cabd0$0501a8c0@windekind.demon.nl> Message-ID: <552640.53858.qm@web34310.mail.mud.yahoo.com> Erwin, If you can give me some details of what you want - IP address of the AsteriskWin32 box - IP address of your ATA - Extension numbers of the phone ports off of your ATA - ATA type I can probably help get you going. You might also want to get a softphone client like XLite or FireFly so you can test getting basic connectivity. John --- Erwin Dokter wrote: > Since Remco is still having problems with his * box, > I decided to play with > Asterisk myself a little. Since I don't have Linux, > I installed > AsteriskWin32, which is a port from Asterisk 1.0.10. > I need a sample > configuration to get it working for CNET. > > The setup is dead simple: ATA > Asterisk > CNET, no > FSX cards and such. So I > need a minimal configuration that excepts calls from > CNET and allows me to > call to CNET. I just activated my office code > (31)318 via email. > > Any help is appriciated. > -- > Erwin Dokter > > > > _______________________________________________ > 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 Jan 3 13:16:14 2007 From: david at josephson.com (David Josephson) Date: Wed, 03 Jan 2007 11:16:14 -0800 Subject: [VoIP] Motorola ATA's, was Re: Update on Linksys Router running Asterisk In-Reply-To: <459B4255.2080105@jps.net> References: <20070103043125.16226.qmail@web34302.mail.mud.yahoo.com> <459B4255.2080105@jps.net> Message-ID: <459C00FE.7080904@josephson.com> Steph Kerman wrote: > I set up a simple network between 2 of my machines and one of the 2 port > Mediatrix units so that I can talk to the Mediatrix unit. The unit has > a configuration interface that you talk to with a browser. This allows > country-specific configuration (call progress tone plan and ringing > frequency) and other characteristics to be set. > > It seems that the Mediatrix APA allows Voip networks to be built using > the APAs themselves and "APA Server software". They seem to be designed > This practice is consistent among early VoIP hardware such as Cisco and 3Com. There was no standards-driven interface (the standards such as H.323 and SCCP were so loosely defined that it was difficult to make interoperable devices without extensive cooperation between vendors) so each maker developed their own. > to function as a seamless extension of the PSTN. The APA FXS ports are > programmed with dialing rules to determine the level of local and toll > access to the PSTN.. The APA FXO ports can be programmed to function > like a DISA, where incoming PSTN callers are answered with dial tone > from the APA, whereupon they can dial any telephone on the Voip > network. They can also be programmed to automatically extend an > incoming PSTN call to any local FXS port on the same APA or any other > APA on the Voip network. > These are common features on most analog telephony adapters. I would strongly suggest that anyone playing with these devices for regular use should stick with units made recently, and which are compliant with the SIP or IAX2 protocols. Older units (I have played with a few, both ATA's and phones) suffer very much from being designed at the foot of the learning curve. Good units are available at around $25 a port or less. voipsupply.com has unlocked Motorola VT1005's in stock (which they have bought from Motorola, and which they claim will continue to be available from them when the present "few hundred" have been sold) for $50 each; these provide two separate FXS ports that will handle DTMF or DP. I have the manuals and provisioning software for these units (for the moment they will email this package to you if you buy some ATA's), email me if you would like to download them (total about 3.6 MB). The provisioning manual is unusually detailed compared to other vendors' and is actually written in English. -- David Josephson From lee at spenadel.com Wed Jan 3 14:07:13 2007 From: lee at spenadel.com (Lee Spenadel) Date: Wed, 3 Jan 2007 15:07:13 -0500 Subject: [VoIP] Last new CNET membber of 2006 In-Reply-To: <459BDB69.9060101@stromberg-carlson.org> Message-ID: <00ac01c72f72$c8b1a110$0ac94da6@Clarabell> Soon come. -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Wednesday, January 03, 2007 11:36 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Last new CNET membber of 2006 SOME of the folks are still doing some testing and making configuration changes. In Lee's case we just got an issue resolved, he needs to relocate his box and connect his switch. Patience is a virtue! John Novack Chad Perkins wrote: > We need to work directory page(s) folks. We continue to get more folks on CNET but if you're not listed no one call you and say hi or listen to your machine(s). > > '363' and '377' hint, hint. > > Chad > > On 1 Jan 2007 at 12:04, John Novack wrote: > >> Lee Spenadel, office code 349, AFAIK was the last new member to join >> CNET in 2006 Last afternoon/evening we got his FXS ATA working over >> CNET Lee and I will be working on more items today and perhaps even >> getting is homebrew SXS switch connected and working, as well as his >> Merlin system >> >> Looking forward to more growth of CNET in 2007 Hope you all have a >> Happy New Year John Novack >> > > > _______________________________________________ > 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 stfkerman at jps.net Wed Jan 3 17:03:41 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 03 Jan 2007 18:03:41 -0500 Subject: [VoIP] Dial Pulse converter specifications In-Reply-To: References: Message-ID: <459C364D.6010401@jps.net> No. It does not prevent dial pulses from reaching the TT device or line. Some ATAs see each dial pulse as a disconnect and reseizure. If the ATA recognized DP correctly, a potential cause of double dialing, you would not need the converter. SK Nathaniel D. Watson wrote: > I do not own one, so I do not know much about it, but would one of the > converters sold by Mike Sandman not serve this purpose? (Provided that the > ATA strictly requires TT dialing, otherwise you'd end up with double-dialing > of each digit). His catalog number for the device is: CID6K. > > Nathan > > > > > on 1/3/07 9:40 AM, Steph Kerman at stfkerman at jps.net wrote: > > >> More loop supervision? What do you mean? >> >> 1) It needs to connect into the loop. A design was published that >> installed in a phone and connected directly to the dial pulse contacts. >> This is not a suitable configuration. >> >> 2) It needs to provide a DC current path during the dial pulse break >> intervals to prevent the downstream device from seeing each pulse as a >> disconnect. >> >> 3) It should support 10 PPS to 20 PPS operation. 20 PPS operation is >> not supported on some modern exchanges that support 10 PPS operation. >> Supporting 20 PPS operation would make it possible to use 20 PPS dials >> both in these environments and in SXS environments equipped with TT >> converters. >> >> 4) It should work with real-world dial pulses that have pulse >> characteristics at the extremes of the readjustment limits published by >> the Bell System. This includes ratio, average and instantaneous speed >> and contact bounce. >> >> 5) A goal would be to make it loop powered, but that's not an absolute >> necessity. It could be designed to only consume power during pulse >> conversion, so that battery operation could be achieved with very long >> battery life. >> >> 6) It should use readily available commercial electronic components: no >> antique switching equipment relays with pure unobtainium contacts. >> >> 7) A printed circuit board would be designed and procured from a >> commercial prototype board house to permit easy assembly without random >> hand wiring. >> >> Steph >> >> john jones wrote: >> >>> Since so many VoIP ATA's don't support DP dialing, >>> there has been a suggestion to construct a TT to DP >>> converter. >>> >>> I'd like to start collecting specifications of what >>> this device needs to do. >>> >>> I expect that on the FXS side, it ought to perform in >>> ways similar to the teltone converters often installed >>> between Linefinders and the 1st selector although I >>> imagine that more loop supervision is required. >>> >>> Let's get a dialog going on this. >>> >>> Thanks! >>> >>> John >>> >>> > > > > > Group web page: http://groups.yahoo.com/group/singingwires > The TCI web site is at http://www.telephonecollectors.org > TCI Picture Place: http://www.telephonecollectors.org/pictures/ > (Picture posting password: tciphotos ) > No password is required to see the pictures. > Yahoo! Groups Links > > <*> To visit your group on the web, go to: > http://groups.yahoo.com/group/singingwires/ > > <*> Your email settings: > Individual Email | Traditional > > <*> To change settings online go to: > http://groups.yahoo.com/group/singingwires/join > (Yahoo! ID required) > > <*> To change settings via email: > mailto:singingwires-digest at yahoogroups.com > mailto:singingwires-fullfeatured at yahoogroups.com > > <*> To unsubscribe from this group, send an email to: > singingwires-unsubscribe at yahoogroups.com > > <*> Your use of Yahoo! Groups is subject to: > http://docs.yahoo.com/info/terms/ > > > From stfkerman at jps.net Wed Jan 3 17:13:00 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 03 Jan 2007 18:13:00 -0500 Subject: [VoIP] Mediatrix APAs In-Reply-To: <459C00FE.7080904@josephson.com> References: <20070103043125.16226.qmail@web34302.mail.mud.yahoo.com> <459B4255.2080105@jps.net> <459C00FE.7080904@josephson.com> Message-ID: <459C387C.5060001@jps.net> More careful reading of the manual states there are H.323, SIP and MGCP versions based on firmware resident in non-volatile but updatable memory. Not sure whether or not that changes anything you said. Steph David Josephson wrote: > Steph Kerman wrote: > >> I set up a simple network between 2 of my machines and one of the 2 port >> Mediatrix units so that I can talk to the Mediatrix unit. The unit has >> a configuration interface that you talk to with a browser. This allows >> country-specific configuration (call progress tone plan and ringing >> frequency) and other characteristics to be set. >> >> It seems that the Mediatrix APA allows Voip networks to be built using >> the APAs themselves and "APA Server software". They seem to be designed >> >> > This practice is consistent among early VoIP hardware such as Cisco and > 3Com. There was no standards-driven interface (the standards such as > H.323 and SCCP were so loosely defined that it was difficult to make > interoperable devices without extensive cooperation between vendors) so > each maker developed their own. > >> to function as a seamless extension of the PSTN. The APA FXS ports are >> programmed with dialing rules to determine the level of local and toll >> access to the PSTN.. The APA FXO ports can be programmed to function >> like a DISA, where incoming PSTN callers are answered with dial tone >> from the APA, whereupon they can dial any telephone on the Voip >> network. They can also be programmed to automatically extend an >> incoming PSTN call to any local FXS port on the same APA or any other >> APA on the Voip network. >> >> > These are common features on most analog telephony adapters. > > I would strongly suggest that anyone playing with these devices for > regular use should stick with units made recently, and which are > compliant with the SIP or IAX2 protocols. Older units (I have played > with a few, both ATA's and phones) suffer very much from being designed > at the foot of the learning curve. Good units are available at around > $25 a port or less. voipsupply.com has unlocked Motorola VT1005's in > stock (which they have bought from Motorola, and which they claim will > continue to be available from them when the present "few hundred" have > been sold) for $50 each; these provide two separate FXS ports that will > handle DTMF or DP. I have the manuals and provisioning software for > these units (for the moment they will email this package to you if you > buy some ATA's), email me if you would like to download them (total > about 3.6 MB). The provisioning manual is unusually detailed compared to > other vendors' and is actually written in English. > > -- > David Josephson > From david at josephson.com Wed Jan 3 18:02:53 2007 From: david at josephson.com (David Josephson) Date: Wed, 03 Jan 2007 16:02:53 -0800 Subject: [VoIP] Mediatrix APAs In-Reply-To: <459C387C.5060001@jps.net> References: <20070103043125.16226.qmail@web34302.mail.mud.yahoo.com> <459B4255.2080105@jps.net> <459C00FE.7080904@josephson.com> <459C387C.5060001@jps.net> Message-ID: <459C442D.3010200@josephson.com> Steph Kerman wrote: > More careful reading of the manual states there are H.323, SIP and MGCP > versions based on firmware resident in non-volatile but updatable > memory. Not sure whether or not that changes anything you said. > Yes, it is common to be able to reload an EEPROM with a different revision for different protocols. For instance I have some very nice Siemens desk phones that can be flashed to be SIP or MGCP. It does not change anything I said -- things that we take for granted, like being able to support STUN (i. e. operate behind NAT) are often missing, and the protocol implementations are often incomplete. My Siemens phones work great -- so long as they are on the same physical network with the switch. Oh, and except that they go to sleep and require a power cycle to start functioning again, after a day or three. "Oh, we fixed that in the next revision of the board and pulled all those from the market." Unless you have a need to share the pain of embedded systems development with some programmers who were just learning how to write microprocessor code, I'd leave them all alone (and I have a nice box of them that are being left alone). -- David From stfkerman at jps.net Wed Jan 3 19:43:32 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 03 Jan 2007 20:43:32 -0500 Subject: [VoIP] Dial Pulse converter specifications In-Reply-To: References: Message-ID: <459C5BC4.6050007@jps.net> Ken, The proposal is to create unTeltone unconverters that do DP->TT rather than TT->DP. The DP counting and DTMF tone generation can be accomplished with a small microcontroller such as a PIC family member. The loop interface of course requires discrete components. There is no need to meet FCC requirement since it will be used in private networks, not directly on the PSTN. Steph kemartinatsnetnet wrote: > Steph, > > So you are basically saying to create something like the Teltone > converters but from the ground up. > > What are you thinking about? Something like a discrete DTMF chip or > build the DTMF front end ground up and control the unit by something > like a Parallax Basic Stamp chip (compiled basic or 'C') which should > be able to store the decoded DTMF and directly control a relay and do > the dial-pulsing. This could get complex if you have to take into > account incoming digit timing, how long to wait for additional > digits, loop supervision, forward disconnect, etc. Plus the actual > hardware design. > > Would this need to meet current FCC connection requirements? > > Questions, questions. > > Ken > > > --- In singingwires at yahoogroups.com, Steph Kerman > wrote: > >> No. It does not prevent dial pulses from reaching the TT device or >> line. Some ATAs see each dial pulse as a disconnect and reseizure. >> >> If the ATA recognized DP correctly, a potential cause of double >> > dialing, > >> you would not need the converter. >> >> SK >> >> Nathaniel D. Watson wrote: >> >>> I do not own one, so I do not know much about it, but would one >>> > of the > >>> converters sold by Mike Sandman not serve this purpose? (Provided >>> > that the > >>> ATA strictly requires TT dialing, otherwise you'd end up with >>> > double-dialing > >>> of each digit). His catalog number for the device is: CID6K. >>> >>> Nathan >>> >>> >>> >>> >>> on 1/3/07 9:40 AM, Steph Kerman at stfkerman at ... wrote: >>> >>> >>> >>>> More loop supervision? What do you mean? >>>> >>>> 1) It needs to connect into the loop. A design was published >>>> > that > >>>> installed in a phone and connected directly to the dial pulse >>>> > contacts. > >>>> This is not a suitable configuration. >>>> >>>> 2) It needs to provide a DC current path during the dial pulse >>>> > break > >>>> intervals to prevent the downstream device from seeing each >>>> > pulse as a > >>>> disconnect. >>>> >>>> 3) It should support 10 PPS to 20 PPS operation. 20 PPS >>>> > operation is > >>>> not supported on some modern exchanges that support 10 PPS >>>> > operation. > >>>> Supporting 20 PPS operation would make it possible to use 20 PPS >>>> > dials > >>>> both in these environments and in SXS environments equipped with >>>> > TT > >>>> converters. >>>> >>>> 4) It should work with real-world dial pulses that have pulse >>>> characteristics at the extremes of the readjustment limits >>>> > published by > >>>> the Bell System. This includes ratio, average and instantaneous >>>> > speed > >>>> and contact bounce. >>>> >>>> 5) A goal would be to make it loop powered, but that's not an >>>> > absolute > >>>> necessity. It could be designed to only consume power during >>>> > pulse > >>>> conversion, so that battery operation could be achieved with >>>> > very long > >>>> battery life. >>>> >>>> 6) It should use readily available commercial electronic >>>> > components: no > >>>> antique switching equipment relays with pure unobtainium >>>> > contacts. > >>>> 7) A printed circuit board would be designed and procured from a >>>> commercial prototype board house to permit easy assembly without >>>> > random > >>>> hand wiring. >>>> >>>> Steph >>>> >>>> john jones wrote: >>>> >>>> >>>>> Since so many VoIP ATA's don't support DP dialing, >>>>> there has been a suggestion to construct a TT to DP >>>>> converter. >>>>> >>>>> I'd like to start collecting specifications of what >>>>> this device needs to do. >>>>> >>>>> I expect that on the FXS side, it ought to perform in >>>>> ways similar to the teltone converters often installed >>>>> between Linefinders and the 1st selector although I >>>>> imagine that more loop supervision is required. >>>>> >>>>> Let's get a dialog going on this. >>>>> >>>>> Thanks! >>>>> >>>>> John >>>>> >>>>> >>>>> >>> >>> >>> Group web page: http://groups.yahoo.com/group/singingwires >>> The TCI web site is at http://www.telephonecollectors.org >>> TCI Picture Place: http://www.telephonecollectors.org/pictures/ >>> (Picture posting password: tciphotos ) >>> No password is required to see the pictures. >>> Yahoo! Groups Links >>> >>> >>> >>> >>> >>> > > > > > Group web page: http://groups.yahoo.com/group/singingwires > The TCI web site is at http://www.telephonecollectors.org > TCI Picture Place: http://www.telephonecollectors.org/pictures/ > (Picture posting password: tciphotos ) > No password is required to see the pictures. > Yahoo! Groups Links > > <*> To visit your group on the web, go to: > http://groups.yahoo.com/group/singingwires/ > > <*> Your email settings: > Individual Email | Traditional > > <*> To change settings online go to: > http://groups.yahoo.com/group/singingwires/join > (Yahoo! ID required) > > <*> To change settings via email: > mailto:singingwires-digest at yahoogroups.com > mailto:singingwires-fullfeatured at yahoogroups.com > > <*> To unsubscribe from this group, send an email to: > singingwires-unsubscribe at yahoogroups.com > > <*> Your use of Yahoo! Groups is subject to: > http://docs.yahoo.com/info/terms/ > > > From stfkerman at jps.net Wed Jan 3 19:53:20 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 03 Jan 2007 20:53:20 -0500 Subject: [VoIP] Mediatrix APAs In-Reply-To: <459C442D.3010200@josephson.com> References: <20070103043125.16226.qmail@web34302.mail.mud.yahoo.com> <459B4255.2080105@jps.net> <459C00FE.7080904@josephson.com> <459C387C.5060001@jps.net> <459C442D.3010200@josephson.com> Message-ID: <459C5E10.1020607@jps.net> Of course it is common to be able to flash new firmware into many or most things made in the last 10 years from modem cards to memory card readers or CDROM drives. My point was with regard to which protocols they support. Do you have a box of the Mediatrix units or something else? As I understand it, the organization they came from has many more still in service and they work. That does not mean they will be useful with * but that does not necessarily make them useful to me. My world is not *-centric. Steph David Josephson wrote: > Steph Kerman wrote: > >> More careful reading of the manual states there are H.323, SIP and MGCP >> versions based on firmware resident in non-volatile but updatable >> memory. Not sure whether or not that changes anything you said. >> >> > Yes, it is common to be able to reload an EEPROM with a different > revision for different protocols. For instance I have some very nice > Siemens desk phones that can be flashed to be SIP or MGCP. It does not > change anything I said -- things that we take for granted, like being > able to support STUN (i. e. operate behind NAT) are often missing, and > the protocol implementations are often incomplete. My Siemens phones > work great -- so long as they are on the same physical network with the > switch. Oh, and except that they go to sleep and require a power cycle > to start functioning again, after a day or three. "Oh, we fixed that in > the next revision of the board and pulled all those from the market." > > Unless you have a need to share the pain of embedded systems development > with some programmers who were just learning how to write microprocessor > code, I'd leave them all alone (and I have a nice box of them that are > being left alone). > > -- > David > > _______________________________________________ > 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 Wed Jan 3 19:48:28 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 03 Jan 2007 20:48:28 -0500 Subject: [VoIP] Dial Pulse converter specifications In-Reply-To: References: Message-ID: <459C5CEC.4030908@jps.net> A circuit board would be designed to permit the circuit to be built up easily from component parts. Designing around a Basic Stamp is a waste of money even for something that will only be built in the dozens. Steph kemartinatsnetnet wrote: > Steph, > > So you are basically saying to create something like the Teltone > converters but from the ground up. > > What are you thinking about? Something like a discrete DTMF chip or > build the DTMF front end ground up and control the unit by something > like a Parallax Basic Stamp chip (compiled basic or 'C') which should > be able to store the decoded DTMF and directly control a relay and do > the dial-pulsing. This could get complex if you have to take into > account incoming digit timing, how long to wait for additional > digits, loop supervision, forward disconnect, etc. Plus the actual > hardware design. > > Would this need to meet current FCC connection requirements? > > Questions, questions. > > Ken > > > --- In singingwires at yahoogroups.com, Steph Kerman > wrote: > >> No. It does not prevent dial pulses from reaching the TT device or >> line. Some ATAs see each dial pulse as a disconnect and reseizure. >> >> If the ATA recognized DP correctly, a potential cause of double >> > dialing, > >> you would not need the converter. >> >> SK >> >> Nathaniel D. Watson wrote: >> >>> I do not own one, so I do not know much about it, but would one >>> > of the > >>> converters sold by Mike Sandman not serve this purpose? (Provided >>> > that the > >>> ATA strictly requires TT dialing, otherwise you'd end up with >>> > double-dialing > >>> of each digit). His catalog number for the device is: CID6K. >>> >>> Nathan >>> >>> >>> >>> >>> on 1/3/07 9:40 AM, Steph Kerman at stfkerman at ... wrote: >>> >>> >>> >>>> More loop supervision? What do you mean? >>>> >>>> 1) It needs to connect into the loop. A design was published >>>> > that > >>>> installed in a phone and connected directly to the dial pulse >>>> > contacts. > >>>> This is not a suitable configuration. >>>> >>>> 2) It needs to provide a DC current path during the dial pulse >>>> > break > >>>> intervals to prevent the downstream device from seeing each >>>> > pulse as a > >>>> disconnect. >>>> >>>> 3) It should support 10 PPS to 20 PPS operation. 20 PPS >>>> > operation is > >>>> not supported on some modern exchanges that support 10 PPS >>>> > operation. > >>>> Supporting 20 PPS operation would make it possible to use 20 PPS >>>> > dials > >>>> both in these environments and in SXS environments equipped with >>>> > TT > >>>> converters. >>>> >>>> 4) It should work with real-world dial pulses that have pulse >>>> characteristics at the extremes of the readjustment limits >>>> > published by > >>>> the Bell System. This includes ratio, average and instantaneous >>>> > speed > >>>> and contact bounce. >>>> >>>> 5) A goal would be to make it loop powered, but that's not an >>>> > absolute > >>>> necessity. It could be designed to only consume power during >>>> > pulse > >>>> conversion, so that battery operation could be achieved with >>>> > very long > >>>> battery life. >>>> >>>> 6) It should use readily available commercial electronic >>>> > components: no > >>>> antique switching equipment relays with pure unobtainium >>>> > contacts. > >>>> 7) A printed circuit board would be designed and procured from a >>>> commercial prototype board house to permit easy assembly without >>>> > random > >>>> hand wiring. >>>> >>>> Steph >>>> >>>> john jones wrote: >>>> >>>> >>>>> Since so many VoIP ATA's don't support DP dialing, >>>>> there has been a suggestion to construct a TT to DP >>>>> converter. >>>>> >>>>> I'd like to start collecting specifications of what >>>>> this device needs to do. >>>>> >>>>> I expect that on the FXS side, it ought to perform in >>>>> ways similar to the teltone converters often installed >>>>> between Linefinders and the 1st selector although I >>>>> imagine that more loop supervision is required. >>>>> >>>>> Let's get a dialog going on this. >>>>> >>>>> Thanks! >>>>> >>>>> John >>>>> >>>>> >>>>> >>> >>> >>> Group web page: http://groups.yahoo.com/group/singingwires >>> The TCI web site is at http://www.telephonecollectors.org >>> TCI Picture Place: http://www.telephonecollectors.org/pictures/ >>> (Picture posting password: tciphotos ) >>> No password is required to see the pictures. >>> Yahoo! Groups Links >>> >>> >>> >>> >>> >>> > > > > > Group web page: http://groups.yahoo.com/group/singingwires > The TCI web site is at http://www.telephonecollectors.org > TCI Picture Place: http://www.telephonecollectors.org/pictures/ > (Picture posting password: tciphotos ) > No password is required to see the pictures. > Yahoo! Groups Links > > <*> To visit your group on the web, go to: > http://groups.yahoo.com/group/singingwires/ > > <*> Your email settings: > Individual Email | Traditional > > <*> To change settings online go to: > http://groups.yahoo.com/group/singingwires/join > (Yahoo! ID required) > > <*> To change settings via email: > mailto:singingwires-digest at yahoogroups.com > mailto:singingwires-fullfeatured at yahoogroups.com > > <*> To unsubscribe from this group, send an email to: > singingwires-unsubscribe at yahoogroups.com > > <*> Your use of Yahoo! Groups is subject to: > http://docs.yahoo.com/info/terms/ > > > From stfkerman at jps.net Wed Jan 3 19:57:38 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 03 Jan 2007 20:57:38 -0500 Subject: [VoIP] Mediatrix APAs, correction In-Reply-To: <459C442D.3010200@josephson.com> References: <20070103043125.16226.qmail@web34302.mail.mud.yahoo.com> <459B4255.2080105@jps.net> <459C00FE.7080904@josephson.com> <459C387C.5060001@jps.net> <459C442D.3010200@josephson.com> Message-ID: <459C5F12.1060105@jps.net> Of course it is common to be able to flash new firmware into many or most things made in the last 10 years from modem cards to memory card readers or CDROM drives. My point was with regard to which protocols they support. Do you have a box of the Mediatrix units or something else? As I understand it, the organization they came from has many more still in service and they work. That does not mean they will be useful with * but that does not necessarily make them useLESS to me. My world is not *-centric. Steph David Josephson wrote: > Steph Kerman wrote: > >> More careful reading of the manual states there are H.323, SIP and MGCP >> versions based on firmware resident in non-volatile but updatable >> memory. Not sure whether or not that changes anything you said. >> >> > Yes, it is common to be able to reload an EEPROM with a different > revision for different protocols. For instance I have some very nice > Siemens desk phones that can be flashed to be SIP or MGCP. It does not > change anything I said -- things that we take for granted, like being > able to support STUN (i. e. operate behind NAT) are often missing, and > the protocol implementations are often incomplete. My Siemens phones > work great -- so long as they are on the same physical network with the > switch. Oh, and except that they go to sleep and require a power cycle > to start functioning again, after a day or three. "Oh, we fixed that in > the next revision of the board and pulled all those from the market." > > Unless you have a need to share the pain of embedded systems development > with some programmers who were just learning how to write microprocessor > code, I'd leave them all alone (and I have a nice box of them that are > being left alone). > > -- > David > > _______________________________________________ > 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 david at josephson.com Wed Jan 3 19:58:24 2007 From: david at josephson.com (David Josephson) Date: Wed, 03 Jan 2007 17:58:24 -0800 Subject: [VoIP] Lucent Merlin expert in the house? Message-ID: <459C5F40.1070707@josephson.com> We got a reasonable deal on a very full Lucent Merlin Legend system with phones and voicemail, for the office here. I have all the docs from the Lucent website and am wading through them. Is there anyone on this list who knows the system well enough to talk me through some of the basic concepts of getting extensions, trunks and ring groups assigned and voice mail running? R7 processor, GS/LS-ID and MLX cards, 007 MLM voicemail. Thanks David Josephson From voiptandem at shaneyoung.com Thu Jan 4 08:12:55 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Thu, 04 Jan 2007 08:12:55 -0600 Subject: [VoIP] ANI Message-ID: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> I'm trying to get a handle on the number of people who have ANI machines on their switches. So far I know Bob Riddell, Phil McCarter and Paul Wills have ANI machines. Does anyone else have one and interested in making it work on CNET? --Shane From jnovack at stromberg-carlson.org Thu Jan 4 08:57:15 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Thu, 04 Jan 2007 09:57:15 -0500 Subject: [VoIP] ANI In-Reply-To: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> References: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> Message-ID: <459D15CB.7020309@stromberg-carlson.org> I WILL have, sometime in the future. I have an ITEC ANI, An E%M card in an Adtran 750 connected to my Asterisk box Once I get the ITEC installed and working, which could take some time Short answer is YES. John Novack Shane Young wrote: > I'm trying to get a handle on the number of people who have ANI > machines on their switches. > > So far I know Bob Riddell, Phil McCarter and Paul Wills have ANI machines. > > Does anyone else have one and interested in making it work on CNET? > > --Shane > > > > _______________________________________________ > 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 Jan 4 12:02:48 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 04 Jan 2007 13:02:48 -0500 Subject: [VoIP] ANI In-Reply-To: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> References: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> Message-ID: <459D4148.2@jps.net> I have a Wescom ANI in storage. Steph Shane Young wrote: > I'm trying to get a handle on the number of people who have ANI machines on their switches. > > So far I know Bob Riddell, Phil McCarter and Paul Wills have ANI machines. > > Does anyone else have one and interested in making it work on CNET? > > --Shane > From voiptandem at shaneyoung.com Thu Jan 4 12:28:46 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Thu, 04 Jan 2007 12:28:46 -0600 Subject: [VoIP] ANI In-Reply-To: <459D4148.2@jps.net> References: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> <459D4148.2@jps.net> Message-ID: <20070104122846.xujbbrjfa8gw0os8@mail.shaneyoung.com> Ok, the current list is: John Novack, Steph Kerman, Bob Riddell, Phil McCarter, Paul Wills From jnovack at stromberg-carlson.org Thu Jan 4 12:41:59 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Thu, 04 Jan 2007 13:41:59 -0500 Subject: [VoIP] ANI In-Reply-To: <20070104122846.xujbbrjfa8gw0os8@mail.shaneyoung.com> References: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> <459D4148.2@jps.net> <20070104122846.xujbbrjfa8gw0os8@mail.shaneyoung.com> Message-ID: <459D4A77.20403@stromberg-carlson.org> Now, can you reveal why you want this information? Should I lock and alarm the doors to my switchroom? Just curious John Novack Shane Young wrote: > Ok, the current list is: > > John Novack, > Steph Kerman, > Bob Riddell, > Phil McCarter, > Paul Wills > > > > > > > _______________________________________________ > 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 Thu Jan 4 14:10:04 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Thu, 04 Jan 2007 14:10:04 -0600 Subject: [VoIP] ANI In-Reply-To: <459D4A77.20403@stromberg-carlson.org> References: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> <459D4148.2@jps.net> <20070104122846.xujbbrjfa8gw0os8@mail.shaneyoung.com> <459D4A77.20403@stromberg-carlson.org> Message-ID: <20070104141004.ag18tklzwwsg8ow8@mail.shaneyoung.com> Yes, but probably not because of me. It helps me understand what the need is to support ANI with other interfaces. Let's say (purely for the sake of argument) that Paul and Steph were the only ones left in the world that wanted to do ANI and didn't have channel banks and T1 cards. If that were the case, I might spend more energy on finding them T1 cards and channel banks rather than making ANI work with other equipment. Quoting John Novack : > Now, can you reveal why you want this information? > Should I lock and alarm the doors to my switchroom? > > Just curious > > John Novack > > > Shane Young wrote: >> Ok, the current list is: >> >> John Novack, >> Steph Kerman, >> Bob Riddell, >> Phil McCarter, >> Paul Wills >> >> >> >> >> >> >> _______________________________________________ >> 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 voiptandem at shaneyoung.com Thu Jan 4 14:13:18 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Thu, 04 Jan 2007 14:13:18 -0600 Subject: [VoIP] ANI In-Reply-To: <20070104141004.ag18tklzwwsg8ow8@mail.shaneyoung.com> References: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> <459D4148.2@jps.net> <20070104122846.xujbbrjfa8gw0os8@mail.shaneyoung.com> <459D4A77.20403@stromberg-carlson.org> <20070104141004.ag18tklzwwsg8ow8@mail.shaneyoung.com> Message-ID: <20070104141318.1fj9h4ixw40oks08@mail.shaneyoung.com> Further, I have only had the opportunity to see an ITT ANI, I would like to see others WORKING :) Quoting Shane Young : > Yes, but probably not because of me. > > It helps me understand what the need is to support ANI with other interfaces. > > Let's say (purely for the sake of argument) that Paul and Steph were > the only ones left in the world that wanted to do ANI and didn't have > channel banks and T1 cards. If that were the case, I might spend more > energy on finding them T1 cards and channel banks rather than making > ANI work with other equipment. > > > > > Quoting John Novack : > >> Now, can you reveal why you want this information? >> Should I lock and alarm the doors to my switchroom? >> >> Just curious >> >> John Novack >> >> >> Shane Young wrote: >>> Ok, the current list is: >>> >>> John Novack, >>> Steph Kerman, >>> Bob Riddell, >>> Phil McCarter, >>> Paul Wills >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> 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/ >> >> > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From jjones3601 at yahoo.com Thu Jan 4 14:17:25 2007 From: jjones3601 at yahoo.com (john jones) Date: Thu, 4 Jan 2007 12:17:25 -0800 (PST) Subject: [VoIP] ANI In-Reply-To: <459D4A77.20403@stromberg-carlson.org> Message-ID: <20070104201725.42633.qmail@web34308.mail.mud.yahoo.com> Be afwaid. Be verwy, verwy afwaid. --- John Novack wrote: > Now, can you reveal why you want this information? > Should I lock and alarm the doors to my switchroom? > > Just curious > > John Novack > > > Shane Young wrote: > > Ok, the current list is: > > > > John Novack, > > Steph Kerman, > > Bob Riddell, > > Phil McCarter, > > Paul Wills > > > > > > > > > > > > > > _______________________________________________ > > 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 jjones3601 at yahoo.com Thu Jan 4 14:19:27 2007 From: jjones3601 at yahoo.com (john jones) Date: Thu, 4 Jan 2007 12:19:27 -0800 (PST) Subject: [VoIP] ANI In-Reply-To: <459D4A77.20403@stromberg-carlson.org> Message-ID: <20070104201927.57638.qmail@web34304.mail.mud.yahoo.com> Gill Smith has one also and Steve Flocke may be able to help get that one working. John --- John Novack wrote: > Now, can you reveal why you want this information? > Should I lock and alarm the doors to my switchroom? > > Just curious > > John Novack > > > Shane Young wrote: > > Ok, the current list is: > > > > John Novack, > > Steph Kerman, > > Bob Riddell, > > Phil McCarter, > > Paul Wills > > > > > > > > > > > > > > _______________________________________________ > > 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 hockd at dteenergy.com Thu Jan 4 14:25:49 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Thu, 4 Jan 2007 15:25:49 -0500 Subject: [VoIP] ANI Message-ID: I'm hunting ANIs Ha Ha Ha HA HA! ;-0 Dennis H. -----voip-bounces at ckts.info wrote: ----- To: jnovack at stromberg-carlson.org, Voice Over IP Tandem for Analog Switches From: john jones Sent by: voip-bounces at ckts.info Date: 01/04/2007 03:17PM Subject: Re: [VoIP] ANI Be afwaid. Be verwy, verwy afwaid. --- John Novack wrote: > Now, can you reveal why you want this information? > Should I lock and alarm the doors to my switchroom? > > Just curious > > John Novack > > > Shane Young wrote: > > Ok, the current list is: > > > > John Novack, > > Steph Kerman, > > Bob Riddell, > > Phil McCarter, > > Paul Wills > > > > > > > > > > > > > > _______________________________________________ > > 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/ > > _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From jnovack at stromberg-carlson.org Thu Jan 4 14:29:39 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Thu, 04 Jan 2007 15:29:39 -0500 Subject: [VoIP] ANI In-Reply-To: <20070104201927.57638.qmail@web34304.mail.mud.yahoo.com> References: <20070104201927.57638.qmail@web34304.mail.mud.yahoo.com> Message-ID: <459D63B3.7020200@stromberg-carlson.org> Correct, though we MAY have to do an upgrade to his machine for a T1. Now if there were an ATA way . . . JN john jones wrote: > Gill Smith has one also and Steve Flocke may be able > to help get that one working. > > John > > > --- John Novack wrote: > > >> Now, can you reveal why you want this information? >> Should I lock and alarm the doors to my switchroom? >> >> Just curious >> >> John Novack >> >> >> Shane Young wrote: >> >>> Ok, the current list is: >>> >>> John Novack, >>> Steph Kerman, >>> Bob Riddell, >>> Phil McCarter, >>> Paul Wills >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> 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 jjones3601 at yahoo.com Thu Jan 4 14:31:18 2007 From: jjones3601 at yahoo.com (john jones) Date: Thu, 4 Jan 2007 12:31:18 -0800 (PST) Subject: [VoIP] ANI In-Reply-To: Message-ID: <495887.67122.qm@web34311.mail.mud.yahoo.com> You crack us up! --- Dennis D Hock wrote: > I'm hunting ANIs Ha Ha Ha HA HA! ;-0 > > Dennis H. > > -----voip-bounces at ckts.info wrote: ----- > > > To: jnovack at stromberg-carlson.org, Voice Over IP > Tandem for Analog Switches > > From: john jones > Sent by: voip-bounces at ckts.info > Date: 01/04/2007 03:17PM > Subject: Re: [VoIP] ANI > > Be afwaid. Be verwy, verwy afwaid. > > > --- John Novack > wrote: > > > Now, can you reveal why you want this information? > > Should I lock and alarm the doors to my > switchroom? > > > > Just curious > > > > John Novack > > > > > > Shane Young wrote: > > > Ok, the current list is: > > > > > > John Novack, > > > Steph Kerman, > > > Bob Riddell, > > > Phil McCarter, > > > Paul Wills > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > 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/ > > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voipProject > 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 pdwills at verizon.net Thu Jan 4 14:47:38 2007 From: pdwills at verizon.net (Paul Wills) Date: Thu, 04 Jan 2007 14:47:38 -0600 (CST) Subject: [VoIP] ANI Message-ID: <22170298.3501941167943661584.JavaMail.root@vms073.mailsrvcs.net> I still have to wonder if, for simplicity's sake, someone in "Asterisk Land" knows how to call the tone detection routines as a freestanding program. The AGI interface can call a program and, in the process, pass it the channel being used and other information. I already figured out how to use the returned info and set a variable when the program is called in the extensions.conf file. I started looking at the AGI function GET DATA which does about 90% of what's needed if only it were possible to get it to work in MF mode. Unfortunately, wherever the tone detection routines are called is too deep for me to find. One benefit of a "freestanding" program is that it should work nicely with an ATA. PDW >From: Shane Young >Date: 2007/01/04 Thu PM 02:10:04 CST >To: voip at ckts.info >Subject: Re: [VoIP] ANI >Yes, but probably not because of me. > >It helps me understand what the need is to support ANI with other interfaces. > >Let's say (purely for the sake of argument) that Paul and Steph were >the only ones left in the world that wanted to do ANI and didn't have >channel banks and T1 cards. If that were the case, I might spend more >energy on finding them T1 cards and channel banks rather than making >ANI work with other equipment. > > > > >Quoting John Novack : > >> Now, can you reveal why you want this information? >> Should I lock and alarm the doors to my switchroom? >> >> Just curious >> >> John Novack >> >> >> Shane Young wrote: >>> Ok, the current list is: >>> >>> John Novack, >>> Steph Kerman, >>> Bob Riddell, >>> Phil McCarter, >>> Paul Wills >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> 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/ >> >> > > > > > >_______________________________________________ >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 Jan 4 14:28:01 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Thu, 04 Jan 2007 15:28:01 -0500 Subject: [VoIP] ANI In-Reply-To: <20070104141318.1fj9h4ixw40oks08@mail.shaneyoung.com> References: <20070104081255.e1we0m8u80ksw0wc@mail.shaneyoung.com> <459D4148.2@jps.net> <20070104122846.xujbbrjfa8gw0os8@mail.shaneyoung.com> <459D4A77.20403@stromberg-carlson.org> <20070104141004.ag18tklzwwsg8ow8@mail.shaneyoung.com> <20070104141318.1fj9h4ixw40oks08@mail.shaneyoung.com> Message-ID: <459D6351.7030407@stromberg-carlson.org> I also have SOME ITT ANI, and possibly later this year "pick up" some more, but am still waiting for docs on that, and can't say if even then I will have enough, not to mention SPACE This ITT was originally configured for a Leich, but in theory that was an add on module, and could be removed. I'll keep you in mind if that comes to pass, but for now the ITEC is the one. Jerry Pettreze also has a T1 and some assorted channel banks, but I can't say if he will ever move towards any ANI If one does enough shopping on eBay, a 24 port FXS equipped channel bank can be had for 100 bucks or so, including shipping. FXO cards can also be had, rarely, for 100 bucks or even a little less. E&M cards are more rare than hen's teeth. I haven't seen one in the last 3-4 months. John Novack Shane Young wrote: > Further, I have only had the opportunity to see an ITT ANI, I would > like to see others WORKING :) > > > > Quoting Shane Young : > > >> Yes, but probably not because of me. >> >> It helps me understand what the need is to support ANI with other interfaces. >> >> Let's say (purely for the sake of argument) that Paul and Steph were >> the only ones left in the world that wanted to do ANI and didn't have >> channel banks and T1 cards. If that were the case, I might spend more >> energy on finding them T1 cards and channel banks rather than making >> ANI work with other equipment. >> >> >> >> >> Quoting John Novack : >> >> >>> Now, can you reveal why you want this information? >>> Should I lock and alarm the doors to my switchroom? >>> >>> Just curious >>> >>> John Novack >>> >>> >>> Shane Young wrote: >>> >>>> Ok, the current list is: >>>> >>>> John Novack, >>>> Steph Kerman, >>>> Bob Riddell, >>>> Phil McCarter, >>>> Paul Wills >>>> >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> 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/ >>> >>> >>> >> >> >> >> _______________________________________________ >> 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 jnovack at stromberg-carlson.org Thu Jan 4 15:32:53 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Thu, 04 Jan 2007 16:32:53 -0500 Subject: [VoIP] ANI In-Reply-To: <22170298.3501941167943661584.JavaMail.root@vms073.mailsrvcs.net> References: <22170298.3501941167943661584.JavaMail.root@vms073.mailsrvcs.net> Message-ID: <459D7285.7030100@stromberg-carlson.org> Sounds as if you are that person! You have progressed much further than I with AGI C code to me looks as if I need to change my baud rate. John Novack Paul Wills wrote: > I still have to wonder if, for simplicity's sake, someone in "Asterisk Land" knows how to call the tone detection routines as a freestanding program. > > The AGI interface can call a program and, in the process, pass it the channel being used and other information. I already figured out how to use the returned info and set a variable when the program is called in the extensions.conf file. I started looking at the AGI function GET DATA which does about 90% of what's needed if only it were possible to get it to work in MF mode. Unfortunately, wherever the tone detection routines are called is too deep for me to find. > > One benefit of a "freestanding" program is that it should work nicely with an ATA. > > PDW > > >> From: Shane Young >> Date: 2007/01/04 Thu PM 02:10:04 CST >> To: voip at ckts.info >> Subject: Re: [VoIP] ANI >> > > >> Yes, but probably not because of me. >> >> It helps me understand what the need is to support ANI with other interfaces. >> >> Let's say (purely for the sake of argument) that Paul and Steph were >> the only ones left in the world that wanted to do ANI and didn't have >> channel banks and T1 cards. If that were the case, I might spend more >> energy on finding them T1 cards and channel banks rather than making >> ANI work with other equipment. >> >> >> >> >> Quoting John Novack : >> >> >>> Now, can you reveal why you want this information? >>> Should I lock and alarm the doors to my switchroom? >>> >>> Just curious >>> >>> John Novack >>> >>> >>> Shane Young wrote: >>> >>>> Ok, the current list is: >>>> >>>> John Novack, >>>> Steph Kerman, >>>> Bob Riddell, >>>> Phil McCarter, >>>> Paul Wills >>>> >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> 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/ >>> >>> >>> >> >> >> >> _______________________________________________ >> 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 voiptandem at shaneyoung.com Thu Jan 4 15:36:20 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Thu, 04 Jan 2007 15:36:20 -0600 Subject: [VoIP] ANI In-Reply-To: <22170298.3501941167943661584.JavaMail.root@vms073.mailsrvcs.net> References: <22170298.3501941167943661584.JavaMail.root@vms073.mailsrvcs.net> Message-ID: <20070104153620.xizbkwg0g8cgswss@mail.shaneyoung.com> Ok, I give up. Here's how you do it your way. :) You need to download and compile "detect" here http://www.phrack.org/archives/50/P50-13 This will decode the MF saved in a wav file. Download this example wav file http://www.shaneyoung.com/kp04886516st.wav # ./detect kp04886516st.wav KP1+04886516+ST Done. Write an AGI that answers and records a file which captures the ANI. Save the file. Run the detect program and parse/save the data returned. Now you have your ANI the way (I think) you wanted to do it with no Channel Bank and no T1 board. Quoting Paul Wills : > I still have to wonder if, for simplicity's sake, someone in > "Asterisk Land" knows how to call the tone detection routines as a > freestanding program. > > The AGI interface can call a program and, in the process, pass it > the channel being used and other information. I already figured out > how to use the returned info and set a variable when the program is > called in the extensions.conf file. I started looking at the AGI > function GET DATA which does about 90% of what's needed if only it > were possible to get it to work in MF mode. Unfortunately, wherever > the tone detection routines are called is too deep for me to find. > > One benefit of a "freestanding" program is that it should work > nicely with an ATA. > > PDW > >> From: Shane Young >> Date: 2007/01/04 Thu PM 02:10:04 CST >> To: voip at ckts.info >> Subject: Re: [VoIP] ANI > >> Yes, but probably not because of me. >> >> It helps me understand what the need is to support ANI with other >> interfaces. >> >> Let's say (purely for the sake of argument) that Paul and Steph were >> the only ones left in the world that wanted to do ANI and didn't have >> channel banks and T1 cards. If that were the case, I might spend more >> energy on finding them T1 cards and channel banks rather than making >> ANI work with other equipment. >> >> >> >> >> Quoting John Novack : >> >>> Now, can you reveal why you want this information? >>> Should I lock and alarm the doors to my switchroom? >>> >>> Just curious >>> >>> John Novack >>> >>> >>> Shane Young wrote: >>>> Ok, the current list is: >>>> >>>> John Novack, >>>> Steph Kerman, >>>> Bob Riddell, >>>> Phil McCarter, >>>> Paul Wills >>>> >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> 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/ >>> >>> >> >> >> >> >> >> _______________________________________________ >> 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 voiptandem at shaneyoung.com Thu Jan 4 15:40:01 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Thu, 04 Jan 2007 15:40:01 -0600 Subject: [VoIP] ANI In-Reply-To: <459D7285.7030100@stromberg-carlson.org> References: <22170298.3501941167943661584.JavaMail.root@vms073.mailsrvcs.net> <459D7285.7030100@stromberg-carlson.org> Message-ID: <20070104154001.es3eqlxyooo0ww48@mail.shaneyoung.com> The nice thing about AGI is you can use ANY language. All of my AGI's are written as bash shell scripts :) Quoting John Novack : > Sounds as if you are that person! > You have progressed much further than I with AGI > C code to me looks as if I need to change my baud rate. > > John Novack > > > Paul Wills wrote: >> I still have to wonder if, for simplicity's sake, someone in >> "Asterisk Land" knows how to call the tone detection routines as a >> freestanding program. >> >> The AGI interface can call a program and, in the process, pass it >> the channel being used and other information. I already figured >> out how to use the returned info and set a variable when the >> program is called in the extensions.conf file. I started looking >> at the AGI function GET DATA which does about 90% of what's needed >> if only it were possible to get it to work in MF mode. >> Unfortunately, wherever the tone detection routines are called is >> too deep for me to find. >> >> One benefit of a "freestanding" program is that it should work >> nicely with an ATA. >> >> PDW >> >> >>> From: Shane Young >>> Date: 2007/01/04 Thu PM 02:10:04 CST >>> To: voip at ckts.info >>> Subject: Re: [VoIP] ANI >>> >> >> >>> Yes, but probably not because of me. >>> >>> It helps me understand what the need is to support ANI with other >>> interfaces. >>> >>> Let's say (purely for the sake of argument) that Paul and Steph were >>> the only ones left in the world that wanted to do ANI and didn't have >>> channel banks and T1 cards. If that were the case, I might spend more >>> energy on finding them T1 cards and channel banks rather than making >>> ANI work with other equipment. >>> >>> >>> >>> >>> Quoting John Novack : >>> >>> >>>> Now, can you reveal why you want this information? >>>> Should I lock and alarm the doors to my switchroom? >>>> >>>> Just curious >>>> >>>> John Novack >>>> >>>> >>>> Shane Young wrote: >>>> >>>>> Ok, the current list is: >>>>> >>>>> John Novack, >>>>> Steph Kerman, >>>>> Bob Riddell, >>>>> Phil McCarter, >>>>> Paul Wills >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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/ >>>> >>>> >>>> >>> >>> >>> >>> _______________________________________________ >>> 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/ >> >> >> > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From pdwills at cedarknolltelephone.com Thu Jan 4 16:14:57 2007 From: pdwills at cedarknolltelephone.com (Paul Wills) Date: Thu, 04 Jan 2007 17:14:57 -0500 Subject: [VoIP] ANI References: <22170298.3501941167943661584.JavaMail.root@vms073.mailsrvcs.net> <20070104153620.xizbkwg0g8cgswss@mail.shaneyoung.com> Message-ID: <001b01c7304d$c890db20$0301a8c0@Main> Wow! That might just do it. PDW ----- Original Message ----- From: "Shane Young" To: Sent: Thursday, January 04, 2007 4:36 PM Subject: Re: [VoIP] ANI > Ok, I give up. Here's how you do it your way. :) > > You need to download and compile "detect" here > http://www.phrack.org/archives/50/P50-13 > > This will decode the MF saved in a wav file. > > > Download this example wav file > http://www.shaneyoung.com/kp04886516st.wav > > # ./detect kp04886516st.wav > KP1+04886516+ST > Done. > > Write an AGI that answers and records a file which captures the ANI. > Save the file. > Run the detect program and parse/save the data returned. > > Now you have your ANI the way (I think) you wanted to do it with no > Channel Bank and no T1 board. > > > > > Quoting Paul Wills : > >> I still have to wonder if, for simplicity's sake, someone in >> "Asterisk Land" knows how to call the tone detection routines as a >> freestanding program. >> >> The AGI interface can call a program and, in the process, pass it >> the channel being used and other information. I already figured out >> how to use the returned info and set a variable when the program is >> called in the extensions.conf file. I started looking at the AGI >> function GET DATA which does about 90% of what's needed if only it >> were possible to get it to work in MF mode. Unfortunately, wherever >> the tone detection routines are called is too deep for me to find. >> >> One benefit of a "freestanding" program is that it should work >> nicely with an ATA. >> >> PDW >> >>> From: Shane Young >>> Date: 2007/01/04 Thu PM 02:10:04 CST >>> To: voip at ckts.info >>> Subject: Re: [VoIP] ANI >> >>> Yes, but probably not because of me. >>> >>> It helps me understand what the need is to support ANI with other >>> interfaces. >>> >>> Let's say (purely for the sake of argument) that Paul and Steph were >>> the only ones left in the world that wanted to do ANI and didn't have >>> channel banks and T1 cards. If that were the case, I might spend more >>> energy on finding them T1 cards and channel banks rather than making >>> ANI work with other equipment. >>> >>> >>> >>> >>> Quoting John Novack : >>> >>>> Now, can you reveal why you want this information? >>>> Should I lock and alarm the doors to my switchroom? >>>> >>>> Just curious >>>> >>>> John Novack >>>> >>>> >>>> Shane Young wrote: >>>>> Ok, the current list is: >>>>> >>>>> John Novack, >>>>> Steph Kerman, >>>>> Bob Riddell, >>>>> Phil McCarter, >>>>> Paul Wills >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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/ >>>> >>>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> 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/ >> >> > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From erwin at windekind.demon.nl Thu Jan 4 16:24:06 2007 From: erwin at windekind.demon.nl (Erwin Dokter) Date: Thu, 4 Jan 2007 23:24:06 +0100 Subject: [VoIP] Configuring Asterisk References: Message-ID: <000d01c7304f$0c4fe260$0501a8c0@windekind.demon.nl> > From: John Novack > > Assumptions: > You have a spare computer, it can boot from CD, and you can burn CD's > from iso files Unfortunately, I don't have a spare, only my W2K server that's already running Web, FTP and mail etc. So no easy way for me to 'just install Linux'. > From: john jones > > Erwin, > > If you can give me some details of what you want > - IP address of the AsteriskWin32 box 192.168.1.3 (external windekind.demon.nl/83.160.231.46) > - IP address of your ATA 192.168.1.9 > - Extension numbers of the phone ports off of your ATA Just one: 31508550255 > - ATA type Sipura 3000, only FSX in use. > I can probably help get you going. Thank you so much! -- Erwin Dokter From stfkerman at jps.net Thu Jan 4 16:33:44 2007 From: stfkerman at jps.net (Steph Kerman) Date: Thu, 04 Jan 2007 17:33:44 -0500 Subject: [VoIP] ANI In-Reply-To: <20070104201927.57638.qmail@web34304.mail.mud.yahoo.com> References: <20070104201927.57638.qmail@web34304.mail.mud.yahoo.com> Message-ID: <459D80C8.9090504@jps.net> I think Rick Walsh has an ANI-D. He'll probably want to use it. Steph john jones wrote: > Gill Smith has one also and Steve Flocke may be able > to help get that one working. > > John > > --- John Novack wrote: >> Shane Young wrote: >> >>> Ok, the current list is: >>> >>> John Novack, >>> Steph Kerman, >>> Bob Riddell, >>> Phil McCarter, >>> Paul Wills > From jnovack at stromberg-carlson.org Thu Jan 4 16:42:10 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Thu, 04 Jan 2007 17:42:10 -0500 Subject: [VoIP] ANI In-Reply-To: <459D80C8.9090504@jps.net> References: <20070104201927.57638.qmail@web34304.mail.mud.yahoo.com> <459D80C8.9090504@jps.net> Message-ID: <459D82C2.4080004@stromberg-carlson.org> He is also looking for a T1 card, but cheap! You may want to contact hims if any turn up He has a D4 channel bank or two, Not sure what cards though. John Novack Steph Kerman wrote: > I think Rick Walsh has an ANI-D. He'll probably want to use it. > > Steph > > john jones wrote: > >> Gill Smith has one also and Steve Flocke may be able >> to help get that one working. >> >> John >> >> --- John Novack wrote: >> >>> Shane Young wrote: >>> >>> >>>> Ok, the current list is: >>>> >>>> John Novack, >>>> Steph Kerman, >>>> Bob Riddell, >>>> Phil McCarter, >>>> Paul Wills >>>> >> >> > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > From wepbx at sbcglobal.net Thu Jan 4 20:33:10 2007 From: wepbx at sbcglobal.net (Richard Walsh) Date: Thu, 4 Jan 2007 18:33:10 -0800 (PST) Subject: [VoIP] ANI In-Reply-To: <20070104122846.xujbbrjfa8gw0os8@mail.shaneyoung.com> Message-ID: <921192.9219.qm@web83314.mail.sp1.yahoo.com> Yes Shane, You can add my name to the list. I have a WECO ANI-D. Rick Walsh Shane Young wrote: Ok, the current list is: John Novack, Steph Kerman, Bob Riddell, Phil McCarter, Paul Wills _______________________________________________ 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 Thu Jan 4 22:30:36 2007 From: greg at vyger.net (Greg Blakely) Date: Thu, 4 Jan 2007 22:30:36 -0600 Subject: [VoIP] Busy Busy Busy Message-ID: Hi, folks. A number of you are waiting on me to register office codes or get logins to the website. And I have to apologize for shining you all on for the last several days. The parts of my life that pay me money have been very, very busy. I will get to you all in the next day or two, I promise. Thanks for your patience. Greg From hockd at dteenergy.com Fri Jan 5 04:02:15 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 5 Jan 2007 05:02:15 -0500 Subject: [VoIP] ANI Message-ID: Glad I can offer a little humor. Things are pretty droll here. Dennis -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: john jones Sent by: voip-bounces at ckts.info Date: 01/04/2007 03:31PM Subject: Re: [VoIP] ANI You crack us up! --- Dennis D Hock wrote: > I'm hunting ANIs Ha Ha Ha HA HA! ;-0 > > Dennis H. > > -----voip-bounces at ckts.info wrote: ----- > > > To: jnovack at stromberg-carlson.org, Voice Over IP > Tandem for Analog Switches > > From: john jones > Sent by: voip-bounces at ckts.info > Date: 01/04/2007 03:17PM > Subject: Re: [VoIP] ANI > > Be afwaid. Be verwy, verwy afwaid. > > > --- John Novack > wrote: > > > Now, can you reveal why you want this information? > > Should I lock and alarm the doors to my > switchroom? > > > > Just curious > > > > John Novack > > > > > > Shane Young wrote: > > > Ok, the current list is: > > > > > > John Novack, > > > Steph Kerman, > > > Bob Riddell, > > > Phil McCarter, > > > Paul Wills > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > 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/ > > > > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voipProject > 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/ > > _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jan 5 04:13:37 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 5 Jan 2007 05:13:37 -0500 Subject: [VoIP] Busy Busy Busy Message-ID: Greg, I am still using the port on my SPA 2K which is hosted to yourself. I have noticed over the holiday perhaps before I can no longer copmplete to anything. Everything gives a fast busy back. When I looked in the web browser it showed the port as registered to the voipvger or something like that. Do you know what may have happened? If we no longer whave the 269-1212 on that port can you assign some number off your machine? I still find it useful to be able to call around the net and back to my * box when I am able to actually work on it. Take care hope you are doing well, Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: From: "Greg Blakely" Sent by: voip-bounces at ckts.info Date: 01/04/2007 11:30PM Subject: [VoIP] Busy Busy Busy Hi, folks. A number of you are waiting on me to register office codes or get logins to the website. And I have to apologize for shining you all on for the last several days. The parts of my life that pay me money have been very, very busy. I will get to you all in the next day or two, I promise. Thanks for your patience. Greg _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jan 5 04:51:10 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 5 Jan 2007 05:51:10 -0500 Subject: [VoIP] Update on Linksys Router running Asterisk Message-ID: John It sounds like you have been very busy in trying to squeeze that load onto the wireless router. It certainly sounds like it may offer an attractive package for some of us to add ourselves or new members to CNET. It is due to the effort of people like you, Greg, John, Shane and all of us together that we are able to progress and learn in a friendly unthreatening or politically driven environment. I myself am still struggling with time and being able to yet get into the * box reliably but John is helping me as he has been doing to several others. Just wanted to say thanks and good luck in moving forward. Thank you, Dennis -----voip-bounces at ckts.info wrote: ----- To: ATCA at LISTSERV.ICORS.ORG, voip at ckts.info, singingwires at yahoogroups.com From: john jones Sent by: voip-bounces at ckts.info Date: 01/02/2007 11:31PM Subject: [VoIP] Update on Linksys Router running Asterisk Good news. I have Asterisk running in a pretty stable mode (actually very stable AFAICT). I have interfaced it to a LinkSys PAP2-NA dual port FXS ATA, a Sipura 3000 with 1 FXS port and 1 FXO port, and a X-Lite softclient running on my laptop. Boy I fought that SPA3K and the working config sure seems like very close to what I started with! I don't think I can run two * boxes on the same (single) dynamic IP address so I'm going to take the existing 688 exchange off the air for a bit and bring up 687 on the Linksys. Eventually, I guess I'll dedicate the Linksys * box to the workshop when the EM swicth is and tandem through the other one. I also have a Cisco ATA that I'm trying to get unlocked and Steph is going to send me a MediaTrix ATA so I'll try to get both of them tested with the Linksys. I'm also going to buy another WRTSL54GS router because I believe that I can VERY easily make a clone of this system and then make minor edits to customize for someone else. I'll make a installation/customization document available along with this clone image if anyone is interested (probably a few weeks out still) John _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From ikj1234i at yahoo.com Sat Jan 6 15:58:57 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Sat, 6 Jan 2007 13:58:57 -0800 (PST) Subject: [VoIP] Call progress sounds In-Reply-To: <45A000A3.4030300@stromberg-carlson.org> Message-ID: <298255.72283.qm@web51604.mail.yahoo.com> Three things... 1. It does seem to be somewhat version dependent, unfortunately. Also, please note that the page at ckts.info, which Greg was kind enough to create, mentions a file download named 'sf.patch' - there were problems (Jon K if I recall correctly) with this version so a newer version was released named 'sf2.patch'. See also the following link http://www.lightlink.com/mhp/sf/sf.txt 2. It would probably actually be quite easy to add code to hear the dial pulses with attenuation (the exact amount might need to be determined experimentally). Apparently my eardrums must be more cast-iron than JN's, since that never bothered me. 3. However what really did bother me was the SxS selector "clunk" sound (the D relay operating to cut thru after the digit) was getting suppressed by the X100P hardware. After all, that particular sound is the one reason why we all have SxS selectors, no? :-) I am hoping that channel bank FXO rather than X100P (or TDM400P) FXO may fix this "problem" , but don't have the equipment here to test*. A separate but related problem was that * doesn't leave very much interdigital time (by default), so the next following digit is starting before you've had time to realize it... * JN - would you be interested in pursuing? Did you say that you had a channel-bank FXO? Max --- John Novack wrote: > This patch was developed for an earlier version of > the Zaptel driver, > and probably is VERY version dependent > Possibly Max Park, who developed it, can comment. > You may have met him > at an Enfield show. > Personally I don't care for it since the dial pulses > are very loud. > Imagine a 302 or 500 where the receiver circuit > wasn't muted. > It would be nice, tho probably not possible, to > mute, but not > completely, the audio. > > JN > > > Lee Spenadel - Computer Medics wrote: > > Does the blip of code on ckts.info work so that > people dialing an SxS > > extension could hear call progress tones? > > > > Here's the link: > > > > http://www.ckts.info/faq.php#10 > > > > > > > > > > > > > > > > > > > > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com From g4vft at btinternet.com Sat Jan 6 17:38:11 2007 From: g4vft at btinternet.com (Jonathan Kay) Date: Sat, 06 Jan 2007 23:38:11 +0000 Subject: [VoIP] Call progress sounds In-Reply-To: <298255.72283.qm@web51604.mail.yahoo.com> References: <298255.72283.qm@web51604.mail.yahoo.com> Message-ID: <45A032E3.9050805@btinternet.com> Yes, I was using Max's sf2 patch successfully on my previous build, which was SVN head from around February 2006. I recently upgraded to 1.2.13, because we were trying to resolve a problem with an ATA, remote hosted off my box. I didn't apply the patch on this occasion. So can't verify that it still works. Two reasons. I did have some comments that it wasn't liked by one caller, mainly because of the loudness of the rotary pulses. Also I'm going to start using my channel bank (when I get time) and didn't think the patch would work with this anyway? The nice feature, was when dialling the Uax12, which is served of level 7 of the Uax13. You could hear the dial tone, which is sent down the trunk. Just as the subscribers would have done, way back when. The other slight pain, was, you had to disable echo-training, if you wanted to hear the first digit out pulse, or most of it gets obscured. I'd be very interested if Max wants to try more refinements and will help out with testing. Jon K From greg at vyger.net Sat Jan 6 19:56:23 2007 From: greg at vyger.net (Greg Blakely) Date: Sat, 6 Jan 2007 19:56:23 -0600 Subject: [VoIP] All-new Hack ! Message-ID: Any changes for Asterisk 1.4? The patch applied well, and zaptel compiled, but asterisk barfed. > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of ikjtel > Sent: Saturday, August 20, 2005 8:18 AM > To: Voice Over IP Tandem for Analog Switches > Subject: [VoIP] All-new Hack ! > > As folks suspected, asterisk silences the audio and doesn't > cut-thru until after dialing is completed. > This behavior is unacceptable to some demented souls... > > This new hack is for anyone in CNET. When an inward call is > placed to your vintage switching office, the caller can now > hear the entire call-setup sequence including > siezure/offhook, dialtone, and dial pulsing. > > Also, I have added support for 2600 Hz inward SF pulsing. If > the caller transmits a 2600 SF tone, the FXO card will send > an onhook signal toward the vintage switching machine for the > duration of the tone. When the tone is removed, the FXO card > will go back offhook. > > Provided that WAN packet loss is low enough*, real-time SF > dial pulsing should be possible. That is, the caller (who > must have SF outpulsing equipment) can direct-dial through > your switch, and hear the individual line finders, selectors, > and connectors operate (assuming an SxS office). The caller > can also disconnect without breaking the VOIP connection. > *- (Also may not work with compressing codecs, etc). > > I would like to ask for beta-testers to help test this patch. > It requires a recompile of asterisk and zaptel, but your > configuration files don't need to be touched. Also, this is > completely unrelated to the x100p "fxs adaptation" hack. > > For more information, please go to > http://www.lightlink.com/mhp/sf/sf.txt > > Max > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection > around http://mail.yahoo.com > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From telesignal at comcast.net Sun Jan 7 10:45:18 2007 From: telesignal at comcast.net (Jeff Seidel) Date: Sun, 07 Jan 2007 11:45:18 -0500 Subject: [VoIP] My Cnet is down Message-ID: <45A1239E.4000605@comcast.net> Hi all, My system is down, someone or some thing changed my root password and I have no idea how to fix it yet. I will let everyone know when Im back up, if I get back up. Jeff Seidel -- Jeff Seidel = E-mail: telesignal at comcast.net See my Railroad Signaling & Communications site at railroad-signaling.com AIM at JSeidel351 From g4vft at btinternet.com Sun Jan 7 10:48:54 2007 From: g4vft at btinternet.com (Jonathan Kay) Date: Sun, 07 Jan 2007 16:48:54 +0000 Subject: [VoIP] My Cnet is down In-Reply-To: <45A1239E.4000605@comcast.net> References: <45A1239E.4000605@comcast.net> Message-ID: <45A12476.6000509@btinternet.com> Jeff Seidel wrote: > Hi all, > > My system is down, someone or some thing changed my root password and I > have no idea how to fix it yet. > > I will let everyone know when Im back up, if I get back up. > > Jeff Seidel > Jeff, You should be able to boot into single user mode, then change the root password, with the passwd comand. http://www.linuxforums.org/security/howto:_recover_root_password.html Jon K From greg at vyger.net Sun Jan 7 12:52:46 2007 From: greg at vyger.net (Greg Blakely) Date: Sun, 7 Jan 2007 12:52:46 -0600 Subject: [VoIP] My Cnet is down, too Message-ID: Well, at least my ATA is down. My main CNET connection is intact. I foolishly set a password in my Sipura 2000, and can't remember it for the life of me. Sipura technical support is more brain dead than I am, claiming that my voip provider must have set a password. In five or six email messages, I've told them at least three times that **I** am my voip provider, and that **I** set the password, and that I am asked for a password before the reset command will occur. Arrrggghhh!!!! I've even taken the sucker apart, and fiddled around with the one strap that is in the beast: various combinations of booting up without the strap in place, then adding it, etc. It's time to bring in the chicken entrails. Either that or resign myself to having a somewhat expensive paperweight. Actually, it's not all that bad. It is set to where the first port goes to icconnecthere.com, and the second one uses an extension off of my asterisk box. As long as I'm content with that config, I'm okay. But I beat myself over the head with whatever heavy object is handy, asking, "How could you have been so dumb?" > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of Jonathan Kay > Sent: Sunday, January 07, 2007 10:49 AM > To: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] My Cnet is down > > Jeff Seidel wrote: > > Hi all, > > > > My system is down, someone or some thing changed my root > password and > > I have no idea how to fix it yet. > > > > I will let everyone know when Im back up, if I get back up. > > > > Jeff Seidel > > > Jeff, > You should be able to boot into single user mode, then change > the root password, with the passwd comand. > http://www.linuxforums.org/security/howto:_recover_root_password.html > > Jon K > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > > From lee at spenadel.com Sun Jan 7 13:18:15 2007 From: lee at spenadel.com (Lee Spenadel) Date: Sun, 7 Jan 2007 14:18:15 -0500 Subject: [VoIP] Office Code 349 Message-ID: <016001c73290$95824350$0ac94da6@Clarabell> Hi everyone, Office code 349 is now online, thanks to the help of John Novack. At this point I have a limited number of extensions connected, as posted on the www.ckts.info website: 349-0911 - My 911 center (yes, you can call this w/o a cruiser making a visit to my house) 349-3960 - My home office Merlin 349-3996 - switch room (not ringing yet) 349-5555 - General announcement - balance of welcome message waiting for the ibmtts site to become available 349-863x - 3 slot payphones connected to my Step switch (there's ten of them) 349-8691 - incoming trunk #1 on 506 manual switchboard 349-85xx - step switch reorder signal 349-8690 - ring, no answer - eventually will be a line appearance on my home office Merlin 349-8698 - 211 space saver on switch frame 349-8699 - step switch busy signal It's exciting to have my stepper online. Any extension, other than my home office line can be called 24/7. Next on the project plan: Add FXS ports so I can have dial 0 in to my 555 as an incoming trunk, as well as removing the 506 from the step and giving it an FXS port. I will be rearranging my step extensions, adding ATA ports, etc. I will also modify the pbx to allow call progress sounds on the step switch. The stepper is a home-built. Three talk paths (two dedicated to POTs, one dedicated to my 3-slots/coin connector). 4 trunks x 20 stations. See ya on CNET. Lee From hockd at dteenergy.com Sun Jan 7 14:32:01 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Sun, 7 Jan 2007 15:32:01 -0500 Subject: [VoIP] My Cnet is down Message-ID: Jeff, Hope it won't be permanent for you. Take care and hope you are having a nice 07 so far. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: voip at ckts.info From: Jeff Seidel Sent by: voip-bounces at ckts.info Date: 01/07/2007 11:45AM Subject: [VoIP] My Cnet is down Hi all, My system is down, someone or some thing changed my root password and I have no idea how to fix it yet. I will let everyone know when Im back up, if I get back up. Jeff Seidel -- Jeff Seidel = E-mail: telesignal at comcast.net See my Railroad Signaling & Communications site at railroad-signaling.com AIM at JSeidel351 _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From g4vft at btinternet.com Sun Jan 7 14:37:14 2007 From: g4vft at btinternet.com (Jonathan Kay) Date: Sun, 07 Jan 2007 20:37:14 +0000 Subject: [VoIP] My Cnet is down, too In-Reply-To: References: Message-ID: <45A159FA.9010602@btinternet.com> Is this any use Greg? To return the SPA-2000 to an unconfigured state, pick up your attached phone handset and type the following: ****73738# Press 1# to confirm. Hang up. The device will reboot with a factory-default configuration. All previously entered configuration settings will be removed. It is possible for service providers to restrict access to this feature. If you are using a SPA that has been provisioned by a provider, you may need the assistance of the service provider in order to perform this action. Jon K From jnovack at stromberg-carlson.org Sun Jan 7 14:37:58 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Sun, 07 Jan 2007 15:37:58 -0500 Subject: [VoIP] Thanks as well to Lee Message-ID: <45A15A26.1050409@stromberg-carlson.org> IN working with Lee, he discovered that SOMEWHERE in a release previous to Asterisk 1.2.13, the ability to wait before dialing pulse was fixed/added Way back when, when this was tried, it only worked in tone dialing out of an FXO port. We all end up helping one another with Asterisk and CNET. With a little luck, Bill Wright will be connected soon For the moment he will be a remote off of Mick Carters Asterisk box, but with his own office code. John Novack From hockd at dteenergy.com Sun Jan 7 14:48:40 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Sun, 7 Jan 2007 15:48:40 -0500 Subject: [VoIP] My Cnet is down, too Message-ID: Thjank Jon You beat me to it. I was going to look that up when I got home from work today. I believe you have it correct. I know at one point very early on I did the same thing to my SPA2K but was able to reset the whole thing and start over. Hopefully Greg will be able to as well. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: Jonathan Kay Sent by: voip-bounces at ckts.info Date: 01/07/2007 03:37PM Subject: Re: [VoIP] My Cnet is down, too Is this any use Greg? To return the SPA-2000 to an unconfigured state, pick up your attached phone handset and type the following: ****73738# Press 1# to confirm. Hang up. The device will reboot with a factory-default configuration. All previously entered configuration settings will be removed. It is possible for service providers to restrict access to this feature. If you are using a SPA that has been provisioned by a provider, you may need the assistance of the service provider in order to perform this action. Jon K _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From voiptandem at shaneyoung.com Sun Jan 7 15:05:34 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Sun, 07 Jan 2007 15:05:34 -0600 Subject: [VoIP] My Cnet is down, too In-Reply-To: References: Message-ID: <20070107150534.y4vw9fpps040cc0k@mail.shaneyoung.com> If the device is locked, you will be prompted for the password at this point. This is to prevent customers from resetting ATA's that belong to service providers. If you buy your own ATA you can enable this feature yourself and you have to remember the password. I beleive this is what Greg has done. I know that we did something like that once and nearly had a dead device. We had to go through our logs of our provisioning software to figure out what a previous password was. If you do it all by hand it's up to you to remember though. Quoting Dennis D Hock : > > Thjank Jon You beat me to it. I was going to look that up when I got home > from work today. I believe you have it correct. I know at one point very > early on I did the same thing to my SPA2K but was able to reset the whole > thing and start over. Hopefully Greg will be able to as well. > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: Jonathan Kay > Sent by: voip-bounces at ckts.info > Date: 01/07/2007 03:37PM > Subject: Re: [VoIP] My Cnet is down, too > > > Is this any use Greg? > > > To return the SPA-2000 to an unconfigured state, pick up your attached > phone handset and type the following: ****73738# Press 1# to confirm. > Hang up. The device will reboot with a factory-default configuration. > All previously entered configuration settings will be removed. It is > possible for service providers to restrict access to this feature. If > you are using a SPA that has been provisioned by a provider, you may > need the assistance of the service provider in order to perform this > action. > > > Jon K > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voipProject 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 telesignal at comcast.net Sun Jan 7 15:38:31 2007 From: telesignal at comcast.net (Jeff Seidel) Date: Sun, 07 Jan 2007 16:38:31 -0500 Subject: [VoIP] Im back up Message-ID: <45A16857.5030705@comcast.net> Hi all, My system is back up and running. Many thanks to Jonathan Kay and others who took the time to help. I appreciate it. Regards, Jeff -- Jeff Seidel = E-mail: telesignal at comcast.net See my Railroad Signaling & Communications site at railroad-signaling.com AIM at JSeidel351 From jnovack at stromberg-carlson.org Sun Jan 7 15:51:47 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Sun, 07 Jan 2007 16:51:47 -0500 Subject: [VoIP] Im back up In-Reply-To: <45A16857.5030705@comcast.net> References: <45A16857.5030705@comcast.net> Message-ID: <45A16B73.2040708@stromberg-carlson.org> Make sure telnet is turned off Port 23 in your router Did someone break in, or was it a "senior moment" ? JN Jeff Seidel wrote: > Hi all, > > My system is back up and running. > > Many thanks to Jonathan Kay and others who took the time to help. I > appreciate it. > > Regards, > Jeff > From ian at uax.org.uk Sun Jan 7 15:53:32 2007 From: ian at uax.org.uk (Ian Jolly) Date: Sun, 7 Jan 2007 21:53:32 -0000 Subject: [VoIP] My Cnet is down, too References: <45A159FA.9010602@btinternet.com> Message-ID: <01a301c732a6$469442f0$0a01a8c0@acer1dd0bbc6d0> My new Linksys 3102 ended up getting locked up and unable to be accessed via the webpage and the resets via the phone didn't work - it ended up being replaced as it was still under guarantee. Ian J. +44 (0)352 82 26 (via a 1929 GPO Rural Automatic eXchange!) CNET - the Heritage Telephone Network ----- Original Message ----- From: "Jonathan Kay" To: "Voice Over IP Tandem for Analog Switches" Sent: Sunday, January 07, 2007 8:37 PM Subject: Re: [VoIP] My Cnet is down, too > > Is this any use Greg? > > > To return the SPA-2000 to an unconfigured state, pick up your attached > phone handset and type the following: ****73738# Press 1# to confirm. > Hang up. The device will reboot with a factory-default configuration. > All previously entered configuration settings will be removed. It is > possible for service providers to restrict access to this feature. If > you are using a SPA that has been provisioned by a provider, you may > need the assistance of the service provider in order to perform this > action. > > > Jon K > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > -- > 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: 7.1.410 / Virus Database: 268.16.7/618 - Release Date: 06/01/2007 > > From stfkerman at jps.net Sun Jan 7 15:53:10 2007 From: stfkerman at jps.net (Steph Kerman) Date: Sun, 07 Jan 2007 16:53:10 -0500 Subject: [VoIP] Some items I'm looking for Message-ID: <45A16BC6.5010607@jps.net> Hi Folks, Here are some things I'm looking for: ------------- Hole blanks or jacks for Lucent Systimax housings - http://www.western-electric.com/Telephony/Station.Equip,.Non-Key/*Misc.Apparatus.Photos/Systimax.Parts.jpg I have some of the housings but no jacks or blanks. The housings normally come with one blank, not enough to fill all or even most holes. They seem to assume that if you chose the 4-port housing you are going to use at least 3 ports or you would have used the 2-port housing. Maybe so. These housings would be great for constructing small special telephone & electronics devices by mounting standard commercial jacks, LEDs or switches in drilled-out blanks. Anyone who uses this product line probably has more blanks than they can possibly use. Does anyone have blanks they do not need? ------------- I need (1) WECo/AT&T, Suttle, ATP or other 104-type dual 8P8C surface mount Cat-3 modular jack. I prefer 110 termination. ------------- Ortronics 4210 - http://www.western-electric.com/Telephony/Station.Equip,.Non-Key/*Misc.Apparatus.Photos/Ortronics.4210.jpg I need some of these Cat-3 multi-jack assemblies. No longer being manufactured. Have not seen any on eBay but I am aware that I might potentially find them there some day. ------------- I'm looking for a Krone wire termination tool or Krone blade for the Harris tool. The Krone tool, IIRC, is grey and shaped like the early 714A tools or Dracon D714 tools... no flange to prevent fingers from slipping onto the terminals. I think it has a little scissor-like cutting blade device at the tip. I would also like to find some Krone test plugs/test cord & plug assemblies. ------------- I'm looking for a Lucent AT8662 "D" Test Cord for 110-type terminals, Comcode 402023956 http://www.western-electric.com/Telephony/Tools,.Test.Sets,.Test.Trunks/Tools.&.Instruments/AT-8662.'D'.Test.Cord.-.3-views.jpg -------------- If anyone has any of these items, please reply with price and condition. Thanks Steph From stfkerman at jps.net Sun Jan 7 15:59:17 2007 From: stfkerman at jps.net (Steph Kerman) Date: Sun, 07 Jan 2007 16:59:17 -0500 Subject: [VoIP] Some items I'm looking for (corrected link) Message-ID: <45A16D35.7010800@jps.net> Hi Folks, Here are some things I'm looking for: ------------- Hole blanks or jacks for Lucent Systimax housings - http://www.western-electric.com/Telephony/Station.Equip,.Non-Key/*Misc.Apparatus.Photos/Systimax.Parts.jpg I have some of the housings but no jacks or blanks. The housings normally come with one blank, not enough to fill all or even most holes. They seem to assume that if you chose the 4-port housing you are going to use at least 3 ports or you would have used the 2-port housing. Maybe so. These housings would be great for constructing small special telephone & electronics devices by mounting standard commercial jacks, LEDs or switches in drilled-out blanks. Anyone who uses this product line probably has more blanks than they can possibly use. Does anyone have blanks they do not need? ------------- I need (1) WECo/AT&T, Suttle, ATP or other 104-type dual 8P8C surface mount Cat-3 modular jack. I prefer 110 termination. ------------- Ortronics 4210 - http://www.western-electric.com/Telephony/Station.Equip,.Non-Key/*Misc.Apparatus.Photos/Ortronics.4210.jpg I need some of these Cat-3 multi-jack assemblies. No longer being manufactured. Have not seen any on eBay but I am aware that I might potentially find them there some day. ------------- I'm looking for a Krone wire termination tool or Krone blade for the Harris tool. The Krone tool, IIRC, is grey and shaped like the early 714A tools or Dracon D714 tools... no flange to prevent fingers from slipping onto the terminals. I think it has a little scissor-like cutting blade device at the tip. I would also like to find some Krone test plugs/test cord & plug assemblies. ------------- I'm looking for a Lucent AT8662 "D" Test Cord for 110-type terminals, Comcode 402023956 http://www.western-electric.com/Telephony/Tools,.Test.Sets,.Test.Trunks/Tools.&.Instruments/AT-8662.D-Type.Test.Cord.-.3-views.jpg -------------- If anyone has any of these items, please reply with price and condition. Thanks Steph From jjones3601 at yahoo.com Sun Jan 7 18:44:10 2007 From: jjones3601 at yahoo.com (john jones) Date: Sun, 7 Jan 2007 16:44:10 -0800 (PST) Subject: [VoIP] MF signalling with Asterisk and update on Linksys platform Message-ID: <20070108004410.89820.qmail@web34309.mail.mud.yahoo.com> My Linksys Asterisk platform seems to be working very well and you can reach me on CNET at 1.687.7700 or 1.687.4878. I have an analog phone hooked up to a Cisco 1750 router with 2 FXS and 2 FXO ports. Some really great news: the 1750 supports DP, MF (at least outbound) and DTMF dialing. I was also able to get a Cisco MC3810 working with my system. These are pretty cool ( and End Of Life ) products. There is a guy on ebay that has 40 of them http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=140071558823 for $129 plus $15 S/H. These units have the max memory, one T1 interface, one Ethernet interface, 4 FXO ports, 2 FXS ports, and 2 serial ports (useless for our purposes). There are E&M modules available from time to time. BTW, I don't know this person but I do have an offer in. Hopefully, next weekend, John Novack, Kirt Stanfield, Paul Wills and I will get together and see if we can get the T1 talking to some channel banks that JN has. If Paul is available, we may try to interface to his system system to try to get his ANI feature working. I can provide configuration templates and samples and assistance to folks if necessary until JN (Mr. CNET) is fully up to speed! I'm interested in learning if others want to try this out. For ~ $250, you can run Asterisk on a Linksys, have (6) analog and digital (1 T1) for about $250. All you need is a broadband connection and some phones. John From ka2wft at arrl.net Sun Jan 7 18:46:13 2007 From: ka2wft at arrl.net (Doug Alderdice) Date: Sun, 07 Jan 2007 19:46:13 -0500 Subject: [VoIP] Motorola VT1005 Message-ID: <5.1.0.14.0.20070107194008.02767888@incoming.verizon.net> I'm intrigued by the availability (and affordability) of the unlocked versions of this unit, per a mention on this list a week or two back, and Voip-Supply.com is almost out my back door here. What's anyone's real-world dial pulse experience with the Motorola VT1005? Is it forgiving of dials that aren't exactly 10 pps, or is the performance along the lines of the Digium FXS card that some have tried with less than stellar results? It would be nice to get around my SmarT-1/Sipura kludge for DP access to CNET. Doug. From stfkerman at jps.net Sun Jan 7 18:56:41 2007 From: stfkerman at jps.net (Steph Kerman) Date: Sun, 07 Jan 2007 19:56:41 -0500 Subject: [VoIP] MF signalling with Asterisk and update on Linksys platform In-Reply-To: <20070108004410.89820.qmail@web34309.mail.mud.yahoo.com> References: <20070108004410.89820.qmail@web34309.mail.mud.yahoo.com> Message-ID: <45A196C9.1010302@jps.net> I don't understand the relationship between the 1750 and 3810. What do the 1750s cost? Which is preferable? By "end of life" I assume you mean no further firmware enhancements or support will be provided. Does this classification apply to both the 1750 and 3810 or only the latter? Steph john jones wrote: > My Linksys Asterisk platform seems to be working very > well and you can reach me on CNET at 1.687.7700 or > 1.687.4878. I have an analog phone hooked up to a > Cisco 1750 router with 2 FXS and 2 FXO ports. > > Some really great news: the 1750 supports DP, MF (at > least outbound) and DTMF dialing. I was also able to > get a Cisco MC3810 working with my system. These are > pretty cool ( and End Of Life ) products. There is a > guy on ebay that has 40 of them > > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=140071558823 > > for $129 plus $15 S/H. > > These units have the max memory, one T1 interface, one > Ethernet interface, 4 FXO ports, 2 FXS ports, and 2 > serial ports (useless for our purposes). There are > E&M modules available from time to time. BTW, I > don't know this person but I do have an offer in. > > Hopefully, next weekend, John Novack, Kirt Stanfield, > Paul Wills and I will get together and see if we can > get the T1 talking to some channel banks that JN has. > If Paul is available, we may try to interface to his > system system to try to get his ANI feature working. > > I can provide configuration templates and samples and > assistance to folks if necessary until JN (Mr. CNET) > is fully up to speed! > > I'm interested in learning if others want to try this > out. For ~ $250, you can run Asterisk on a Linksys, > have (6) analog and digital (1 T1) for about $250. > All you need is a broadband connection and some > phones. > > John > > From greg at vyger.net Sun Jan 7 19:04:40 2007 From: greg at vyger.net (Greg Blakely) Date: Sun, 7 Jan 2007 19:04:40 -0600 Subject: [VoIP] My Cnet is down, too Message-ID: Yeah, that's what I was afraid of. > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of Shane Young > Sent: Sunday, January 07, 2007 3:06 PM > To: voip at ckts.info > Subject: Re: [VoIP] My Cnet is down, too > > If the device is locked, you will be prompted for the > password at this point. > > > This is to prevent customers from resetting ATA's that belong > to service providers. > > If you buy your own ATA you can enable this feature yourself > and you have to remember the password. I beleive this is > what Greg has done. > > I know that we did something like that once and nearly had a > dead device. We had to go through our logs of our > provisioning software to figure out what a previous password was. > > If you do it all by hand it's up to you to remember though. > > > Quoting Dennis D Hock : > > > > > Thjank Jon You beat me to it. I was going to look that up > when I got > > home from work today. I believe you have it correct. I > know at one > > point very early on I did the same thing to my SPA2K but > was able to > > reset the whole thing and start over. Hopefully Greg will > be able to as well. > > > > Dennis Hock > > > > -----voip-bounces at ckts.info wrote: ----- > > > > > > To: Voice Over IP Tandem for Analog Switches > > From: Jonathan Kay Sent by: > > voip-bounces at ckts.info > > Date: 01/07/2007 03:37PM > > Subject: Re: [VoIP] My Cnet is down, too > > > > > > Is this any use Greg? > > > > > > To return the SPA-2000 to an unconfigured state, pick up > your attached > > phone handset and type the following: ****73738# Press 1# > to confirm. > > Hang up. The device will reboot with a factory-default > configuration. > > All previously entered configuration settings will be > removed. It is > > possible for service providers to restrict access to this > feature. If > > you are using a SPA that has been provisioned by a > provider, you may > > need the assistance of the service provider in order to > perform this > > action. > > > > > > Jon K > > > > _______________________________________________ > > VoIP mailing list > > VoIP at ckts.info > > http://lists.ckts.info/mailman/listinfo/voipProject Web Page: > > http://www.ckts.info/ > > _______________________________________________ > > VoIP mailing list > > VoIP at ckts.info > > http://lists.ckts.info/mailman/listinfo/voip > > Projec