From pdwills at cedarknolltelephone.com Tue May 1 19:51:21 2007 From: pdwills at cedarknolltelephone.com (Paul Wills) Date: Tue, 01 May 2007 20:51:21 -0400 Subject: [VoIP] Missed Calls References: <4616442A.3070008@fubegra.net> <461659C0.5000200@jps.net> <4616D743.40707@fubegra.net> Message-ID: <008a01c78c54$026fc790$0301a8c0@Main> Hello all, I'm still not getting some random calls. It appears that if I call another CNET exchange and they call back, they connect but if someone tries calling first, they get nothing. I see no sign of an attempt if I watch the Asterisk CLI. Any thoughts? PDW From jnovack at stromberg-carlson.org Tue May 1 20:01:07 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Tue, 01 May 2007 21:01:07 -0400 Subject: [VoIP] Missed Calls In-Reply-To: <008a01c78c54$026fc790$0301a8c0@Main> References: <4616442A.3070008@fubegra.net> <461659C0.5000200@jps.net> <4616D743.40707@fubegra.net> <008a01c78c54$026fc790$0301a8c0@Main> Message-ID: <4637E2D3.7000105@stromberg-carlson.org> I just tried to call what I believe is one of your working numbers and got this: -- Executing Dial("Zap/24-1", "IAX2/cnetguest at cedartel.homedns.org/15943516") in new stack -- Called cnetguest at cedartel.homedns.org/15943516 May 1 20:58:37 WARNING[1819]: chan_iax2.c:1767 attempt_transmit: Max retries exceeded to host 71.162.149.32 on IAX2/71.162.149.32:4569-4 (type = 6, subclass = 1, ts=11, seqno=0) -- Hungup 'IAX2/71.162.149.32:4569-4' Can't say if that will help or not. Is that your current IP address?? John Novack Paul Wills wrote: > Hello all, > > I'm still not getting some random calls. It appears that if I call another > CNET exchange and they call back, they connect but if someone tries calling > first, they get nothing. I see no sign of an attempt if I watch the > Asterisk CLI. > > Any thoughts? > > PDW > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From ka2wft at arrl.net Tue May 1 20:30:15 2007 From: ka2wft at arrl.net (Doug Alderdice) Date: Tue, 01 May 2007 21:30:15 -0400 Subject: [VoIP] Missed Calls In-Reply-To: <008a01c78c54$026fc790$0301a8c0@Main> References: <4616442A.3070008@fubegra.net> <461659C0.5000200@jps.net> <4616D743.40707@fubegra.net> Message-ID: <5.1.0.14.2.20070501212440.0288db78@incoming.verizon.net> At 08:51 PM 5/1/2007 -0400, Paul Wills wrote: >I'm still not getting some random calls. It appears that if I call another >CNET exchange and they call back, they connect but if someone tries calling >first, they get nothing. I see no sign of an attempt if I watch the >Asterisk CLI. It almost sounds to me like a router issue. You make a call from your system and the router "remembers" it's there. After a while, it "forgets." Are you running * within your router? Seems like it should always know it's there if you are. If it were a problem on my system, I think I'd poke around in the router settings and see if anything is amiss. Doug. From lee at spenadel.com Tue May 1 22:26:34 2007 From: lee at spenadel.com (Lee Spenadel) Date: Tue, 1 May 2007 23:26:34 -0400 Subject: [VoIP] Missed Calls In-Reply-To: <5.1.0.14.2.20070501212440.0288db78@incoming.verizon.net> References: <4616442A.3070008@fubegra.net> <461659C0.5000200@jps.net> <4616D743.40707@fubegra.net> <5.1.0.14.2.20070501212440.0288db78@incoming.verizon.net> Message-ID: <013a01c78c69$afbb4950$0f31dbf0$@com> Assuming this your average "run-of-the-mill" consumer router, they don't remember anything. If it has been provisioned with the default routing settings, then all it's looking for is the default gateway, or the "next hop" in the outbound network routing chain. The default gateway is the ISPs router that is the next hop. That router then forwards it to the next hop based on what the routing tables are telling it is the best route. Consumer routers don't run routing algorithms, such as RIP or OSPF unless you program it to do so. The default is to send the packet to the next hop, or the default gateway. The routers on the Internet are a completely different story and they do cache the routes to other IP networks. I would try some basic network diagnostics, such as seeing if someone can do a consistent tracert to your IP. It appears that the problem is a routing issue to your IP. It could also be that your IP is changing rapidly. I had this problem with a customer who had remote server access pegged to a dynamic dns service. Verizon was bouncing their IP address almost every 2 minutes, making it impossible to track the updates through dynamic dns services because that service could not be updated sooner than every 15 minutes. So I would ensure that your IP is not changing repeatedly and I would check to make sure that your IP network is reachable on a consistent basis. Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of Doug Alderdice Sent: Tuesday, May 01, 2007 9:30 PM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Missed Calls At 08:51 PM 5/1/2007 -0400, Paul Wills wrote: >I'm still not getting some random calls. It appears that if I call another >CNET exchange and they call back, they connect but if someone tries calling >first, they get nothing. I see no sign of an attempt if I watch the >Asterisk CLI. It almost sounds to me like a router issue. You make a call from your system and the router "remembers" it's there. After a while, it "forgets." Are you running * within your router? Seems like it should always know it's there if you are. If it were a problem on my system, I think I'd poke around in the router settings and see if anything is amiss. Doug. _______________________________________________ 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 Wed May 2 04:41:47 2007 From: pdwills at cedarknolltelephone.com (Paul Wills) Date: Wed, 02 May 2007 05:41:47 -0400 Subject: [VoIP] Missed Calls References: <4616442A.3070008@fubegra.net> <461659C0.5000200@jps.net> <4616D743.40707@fubegra.net> <008a01c78c54$026fc790$0301a8c0@Main> <4637E2D3.7000105@stromberg-carlson.org> Message-ID: <003801c78c9f$aee57330$0301a8c0@Main> As of this morning (5:41 AM EDT) it is. PDW ----- Original Message ----- From: "John Novack" To: "Voice Over IP Tandem for Analog Switches" Sent: Tuesday, May 01, 2007 9:01 PM Subject: Re: [VoIP] Missed Calls >I just tried to call what I believe is one of your working numbers and > got this: > > -- Executing Dial("Zap/24-1", > "IAX2/cnetguest at cedartel.homedns.org/15943516") in new stack > -- Called cnetguest at cedartel.homedns.org/15943516 > May 1 20:58:37 WARNING[1819]: chan_iax2.c:1767 attempt_transmit: Max > retries exceeded to host 71.162.149.32 on IAX2/71.162.149.32:4569-4 > (type = 6, subclass = 1, ts=11, seqno=0) > -- Hungup 'IAX2/71.162.149.32:4569-4' > > Can't say if that will help or not. > Is that your current IP address?? > > John Novack > > > Paul Wills wrote: >> Hello all, >> >> I'm still not getting some random calls. It appears that if I call >> another >> CNET exchange and they call back, they connect but if someone tries >> calling >> first, they get nothing. I see no sign of an attempt if I watch the >> Asterisk CLI. >> >> Any thoughts? >> >> PDW >> From pdwills at cedarknolltelephone.com Wed May 2 04:45:13 2007 From: pdwills at cedarknolltelephone.com (Paul Wills) Date: Wed, 02 May 2007 05:45:13 -0400 Subject: [VoIP] Missed Calls References: <4616442A.3070008@fubegra.net> <461659C0.5000200@jps.net> <4616D743.40707@fubegra.net> <5.1.0.14.2.20070501212440.0288db78@incoming.verizon.net> Message-ID: <003901c78c9f$af1053c0$0301a8c0@Main> I'm not running * in my "main" router. As far as I know, the * router is merely being used as a switch but I will double check the settings. I'm not even using the WAN port. PDW ----- Original Message ----- From: "Doug Alderdice" To: "Voice Over IP Tandem for Analog Switches" Sent: Tuesday, May 01, 2007 9:30 PM Subject: Re: [VoIP] Missed Calls > > At 08:51 PM 5/1/2007 -0400, Paul Wills wrote: >>I'm still not getting some random calls. It appears that if I call >>another >>CNET exchange and they call back, they connect but if someone tries >>calling >>first, they get nothing. I see no sign of an attempt if I watch the >>Asterisk CLI. > > It almost sounds to me like a router issue. You make a call from your > system and the router "remembers" it's there. After a while, it > "forgets." Are you running * within your router? Seems like it should > always know it's there if you are. > > If it were a problem on my system, I think I'd poke around in the router > settings and see if anything is amiss. > > Doug. > > From pdwills at verizon.net Wed May 2 07:02:10 2007 From: pdwills at verizon.net (Paul Wills) Date: Wed, 02 May 2007 07:02:10 -0500 (CDT) Subject: [VoIP] Missed Calls Message-ID: <26000354.641351178107330533.JavaMail.root@vms226.mailsrvcs.net> I just successfully connected to the * switch via SSH from the office. Thought: I wonder if I have the proper range of ports defined in the port forwarding. Could this be a cause for missed calls? What is the proper range? PDW From lee at spenadel.com Wed May 2 08:08:48 2007 From: lee at spenadel.com (Lee Spenadel) Date: Wed, 2 May 2007 09:08:48 -0400 Subject: [VoIP] Missed Calls In-Reply-To: <26000354.641351178107330533.JavaMail.root@vms226.mailsrvcs.net> References: <26000354.641351178107330533.JavaMail.root@vms226.mailsrvcs.net> Message-ID: <016b01c78cbb$059e1e50$10da5af0$@com> I'd turn off the Linux firewall, unless you have multiple LAN/WAN segments attached to it. Use the public facing firewall (router that faces your ISP) to mange access to Asterisk is easier. The ports you need are: 5060 - SIP (I forward TCP/UDP since I can't remember which it is) 4569 - IAX2 - UDP 10000 - 20000 - RTP - UDP 5036 - IAX - UDP -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of Paul Wills Sent: Wednesday, May 02, 2007 8:02 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Missed Calls I just successfully connected to the * switch via SSH from the office. Thought: I wonder if I have the proper range of ports defined in the port forwarding. Could this be a cause for missed calls? What is the proper range? PDW _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From john_reads_cnet_via_archives at covert.org Thu May 3 12:19:50 2007 From: john_reads_cnet_via_archives at covert.org (John R. Covert) Date: Thu, 3 May 2007 13:19:50 -0400 (EDT) Subject: [VoIP] Missed Calls Message-ID: <20070503172641.19F005630F@ns1.vyger.net> The behaviour Paul describes is exactly what would happen if his IAX2 port isn't forwarded. Callers would attempt to connect to port 4569 at his firewall, which would ignore all calls (that explains the "Max retries exceeded" message John Novack reported). However, if there had recently been an outbound IAX2 call to the address of another CNET member, when that member calls back, the firewall would still have the IAX2 port open. Lee wrote: >5060 - SIP (I forward TCP/UDP since I can't remember which it is) SIP can be done with either, but it's universally done only with UDP. CNET doesn't use SIP. >4569 - IAX2 - UDP This is the main requirement for CNET. >10000 - 20000 - RTP - UDP But only if using SIP. On IAX2 everything goes on 4569. >5036 - IAX - UDP No one uses this anymore. /john From pdwills at verizon.net Thu May 3 12:46:56 2007 From: pdwills at verizon.net (Paul Wills) Date: Thu, 03 May 2007 12:46:56 -0500 (CDT) Subject: [VoIP] Missed Calls Message-ID: <2020561.120921178214417068.JavaMail.root@vms068.mailsrvcs.net> John, Thanks. That was definitely missing. I added the port but need a few "cold calls" to test it. (If you try it before 5:00 PM, don't call a human as my wife doesn't usually answer CNET calls.) PDW >From: "John R. Covert" >Date: 2007/05/03 Thu PM 12:19:50 CDT >To: CNET >Subject: [VoIP] Missed Calls >The behaviour Paul describes is exactly what would happen if his >IAX2 port isn't forwarded. Callers would attempt to connect to >port 4569 at his firewall, which would ignore all calls (that >explains the "Max retries exceeded" message John Novack reported). > >However, if there had recently been an outbound IAX2 call to the >address of another CNET member, when that member calls back, the >firewall would still have the IAX2 port open. > >Lee wrote: > >>5060 - SIP (I forward TCP/UDP since I can't remember which it is) > >SIP can be done with either, but it's universally done only with UDP. >CNET doesn't use SIP. > >>4569 - IAX2 - UDP > >This is the main requirement for CNET. > >>10000 - 20000 - RTP - UDP > >But only if using SIP. On IAX2 everything goes on 4569. > >>5036 - IAX - UDP > >No one uses this anymore. > >/john >_______________________________________________ >VoIP mailing list >VoIP at ckts.info >http://lists.ckts.info/mailman/listinfo/voip >Project Web Page: http://www.ckts.info/ From ka2wft at arrl.net Thu May 3 14:59:28 2007 From: ka2wft at arrl.net (Doug Alderdice) Date: Thu, 03 May 2007 15:59:28 -0400 Subject: [VoIP] Missed Calls In-Reply-To: <2020561.120921178214417068.JavaMail.root@vms068.mailsrvcs. net> Message-ID: <5.1.0.14.2.20070503155902.00bc8990@incoming.verizon.net> At 12:46 PM 5/3/2007 -0500, you wrote: >John, > >Thanks. That was definitely missing. I added the port but need a few >"cold calls" to test it. (If you try it before 5:00 PM, don't call a >human as my wife doesn't usually answer CNET calls.) > >PDW Milliwatt line worked fine from the 366/833 exchange just now. D. > >From: "John R. Covert" > >Date: 2007/05/03 Thu PM 12:19:50 CDT > >To: CNET > >Subject: [VoIP] Missed Calls > > >The behaviour Paul describes is exactly what would happen if his > >IAX2 port isn't forwarded. Callers would attempt to connect to > >port 4569 at his firewall, which would ignore all calls (that > >explains the "Max retries exceeded" message John Novack reported). > > > >However, if there had recently been an outbound IAX2 call to the > >address of another CNET member, when that member calls back, the > >firewall would still have the IAX2 port open. > > > >Lee wrote: > > > >>5060 - SIP (I forward TCP/UDP since I can't remember which it is) > > > >SIP can be done with either, but it's universally done only with UDP. > >CNET doesn't use SIP. > > > >>4569 - IAX2 - UDP > > > >This is the main requirement for CNET. > > > >>10000 - 20000 - RTP - UDP > > > >But only if using SIP. On IAX2 everything goes on 4569. > > > >>5036 - IAX - UDP > > > >No one uses this anymore. > > > >/john From t2600 at sbcglobal.net Thu May 3 23:21:46 2007 From: t2600 at sbcglobal.net (Jerry Petrizze) Date: Thu, 3 May 2007 21:21:46 -0700 Subject: [VoIP] AE Connector Message-ID: <00cb01c78e03$bac8db00$0301a8c0@pentium4> Might anyone have the schematic of an AE 236636-R connector that would be willing to make a copy? Thank you Jerry Petrizze From greg at vyger.net Wed May 9 21:03:03 2007 From: greg at vyger.net (Greg Blakely) Date: Wed, 9 May 2007 21:03:03 -0500 Subject: [VoIP] 1-222 available Message-ID: I've decided to give up +1-222. It really has no historical meaning for me, so I'm going to just go with +1-499 for my analog switch, and +1-652 for my asterisk-run fxs phones. It's now warm enough to get out to work on my SC-30R, so there might actually BE something on +1-499 (again) soon. From jnovack at stromberg-carlson.org Fri May 11 21:24:45 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Fri, 11 May 2007 22:24:45 -0400 Subject: [VoIP] CON cast Message-ID: <4645256D.6020901@stromberg-carlson.org> CON cast is at it again. They seem to be on a mission to change my IP address and a short DHCP lease time, then my Linksys router fails to ask for a renewal, so all goes dead until I goose it from the setup screens. They have gone through this before, then all settles down for a while. I suppose their IT boys are up to something, but it sure is a pain. If you find my CNET connection down at a given time, that is probably the reason Wonder if I went to another brand of router things would improve? John Novack From lee at spenadel.com Sat May 12 06:55:57 2007 From: lee at spenadel.com (Lee Spenadel) Date: Sat, 12 May 2007 07:55:57 -0400 Subject: [VoIP] CON cast In-Reply-To: <4645256D.6020901@stromberg-carlson.org> References: <4645256D.6020901@stromberg-carlson.org> Message-ID: <022b01c7948c$8184e110$848ea330$@com> John, You don't need to go to the setup screen - just power cycle the router by unplugging the power connector from the back of it. I see this problem with customers. A firmware update to the router may help. It could be the router itself. Other brands of consumer grade routers do this. I rarely see this behavior with more expensive routers like those from sonic. Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Friday, May 11, 2007 10:25 PM To: Voice Over IP Subject: [VoIP] CON cast CON cast is at it again. They seem to be on a mission to change my IP address and a short DHCP lease time, then my Linksys router fails to ask for a renewal, so all goes dead until I goose it from the setup screens. They have gone through this before, then all settles down for a while. I suppose their IT boys are up to something, but it sure is a pain. If you find my CNET connection down at a given time, that is probably the reason Wonder if I went to another brand of router things would improve? John Novack _______________________________________________ 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 Sat May 12 13:10:02 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Sat, 12 May 2007 14:10:02 -0400 Subject: [VoIP] CON cast In-Reply-To: <022b01c7948c$8184e110$848ea330$@com> References: <4645256D.6020901@stromberg-carlson.org> <022b01c7948c$8184e110$848ea330$@com> Message-ID: <464602FA.4090005@stromberg-carlson.org> Lee Spenadel wrote: > John, > > You don't need to go to the setup screen - just power cycle the router by unplugging the power connector from the back of it. In my case it is as easy to do a DHCP release and renew > I see this problem with customers. A firmware update to the router may help. Doesn't seem to be available. In addition, this from Linksys: Based on customer feedback, North America E-mail support will be discontinued starting May 15th, 2007. E-mail received before that date will be handled and responses will be sent out until June 1st, when the email system will be shut down completely. > It could be the router itself. Only seems to happen when CON cast places me in a specific IP range with a short lease time!! > Other brands of consumer grade routers do this. I rarely > see this behavior with more expensive routers like those from sonic. > I have a used SonicWall router here, but have yet to see if I can set it up I also have a computer with Monowall installed, but couldn't figure out how to open up the port for CNET. I SUPPOSE I could also use my Vonage V1005V. Or simply tough it through until CONCAST stops playing games. JN > Lee > > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of > John Novack > Sent: Friday, May 11, 2007 10:25 PM > To: Voice Over IP > Subject: [VoIP] CON cast > > CON cast is at it again. They seem to be on a mission to change my IP > address and a short DHCP lease time, then my Linksys router fails to ask > for a renewal, so all goes dead until I goose it from the setup screens. > They have gone through this before, then all settles down for a while. > I suppose their IT boys are up to something, but it sure is a pain. > If you find my CNET connection down at a given time, that is probably > the reason > > Wonder if I went to another brand of router things would improve? > > John Novack > _______________________________________________ > 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 Sat May 12 17:59:13 2007 From: greg at vyger.net (Greg Blakely) Date: Sat, 12 May 2007 17:59:13 -0500 Subject: [VoIP] CON cast Message-ID: When an IP address is handed out via DHCP, the server notifies the client how long the lease is. At somewhere around halfway to the expiration, the client *should* start asking the server for a renewal. If the address is available, a new lease is agreed upon, and the clocks reset. If, however, Comcast simply shut down one server and fired up another, then, as Lee suggested, your best bet would be to simply power cycle your router. It will then re-discover (or discover for the first time) the DHCP server on your subnet. On the other hand, if Comcast's servers haven't changed, your router, if it is RFC compliant, will ask for a new address 1/2-way through its lease. If not, it's not really Comcast that is at fault. > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of John Novack > Sent: Saturday, May 12, 2007 1:10 PM > To: Lee Spenadel > Cc: 'Voice Over IP Tandem for Analog Switches' > Subject: Re: [VoIP] CON cast > > > > Lee Spenadel wrote: > > John, > > > > You don't need to go to the setup screen - just power cycle > the router by unplugging the power connector from the back of it. > In my case it is as easy to do a DHCP release and renew > > I see this problem with customers. A firmware update to > the router may help. > Doesn't seem to be available. > In addition, this from Linksys: > > Based on customer feedback, North America E-mail support will > be discontinued starting May 15th, 2007. E-mail received > before that date will be handled and responses will be sent > out until June 1st, when the email system will be shut down > completely. > > It could be the router itself. > Only seems to happen when CON cast places me in a specific IP > range with a short lease time!! > > Other brands of consumer grade routers do this. I rarely see this > > behavior with more expensive routers like those from sonic. > > > I have a used SonicWall router here, but have yet to see if I > can set it up I also have a computer with Monowall installed, > but couldn't figure out how to open up the port for CNET. > I SUPPOSE I could also use my Vonage V1005V. > Or simply tough it through until CONCAST stops playing games. > > JN > > > Lee > > > > -----Original Message----- > > From: voip-bounces at ckts.info > [mailto:voip-bounces at ckts.info] On Behalf > > Of John Novack > > Sent: Friday, May 11, 2007 10:25 PM > > To: Voice Over IP > > Subject: [VoIP] CON cast > > > > CON cast is at it again. They seem to be on a mission to > change my IP > > address and a short DHCP lease time, then my Linksys router > fails to > > ask for a renewal, so all goes dead until I goose it from > the setup screens. > > They have gone through this before, then all settles down > for a while. > > I suppose their IT boys are up to something, but it sure is a pain. > > If you find my CNET connection down at a given time, that > is probably > > the reason > > > > Wonder if I went to another brand of router things would improve? > > > > 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 lee at spenadel.com Sat May 12 19:45:55 2007 From: lee at spenadel.com (Lee Spenadel) Date: Sat, 12 May 2007 20:45:55 -0400 Subject: [VoIP] CON cast In-Reply-To: References: Message-ID: <026801c794f8$11247270$336d5750$@com> Greg, you are correct. However, I've seen all too often that when Verizon or ConCast (here on the Cape) change the IP address (through lease expiration or subnet change on their end) that consumer grade routers like Linksys 4 port router/switch do not pick up the new address. They sit there in a state where it can't renew its DHCP lease. It's a standards compliant router with bugs. So in theory it should release/renew properly, but we all know how that goes...... -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of Greg Blakely Sent: Saturday, May 12, 2007 6:59 PM To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] CON cast When an IP address is handed out via DHCP, the server notifies the client how long the lease is. At somewhere around halfway to the expiration, the client *should* start asking the server for a renewal. If the address is available, a new lease is agreed upon, and the clocks reset. If, however, Comcast simply shut down one server and fired up another, then, as Lee suggested, your best bet would be to simply power cycle your router. It will then re-discover (or discover for the first time) the DHCP server on your subnet. On the other hand, if Comcast's servers haven't changed, your router, if it is RFC compliant, will ask for a new address 1/2-way through its lease. If not, it's not really Comcast that is at fault. > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of John Novack > Sent: Saturday, May 12, 2007 1:10 PM > To: Lee Spenadel > Cc: 'Voice Over IP Tandem for Analog Switches' > Subject: Re: [VoIP] CON cast > > > > Lee Spenadel wrote: > > John, > > > > You don't need to go to the setup screen - just power cycle > the router by unplugging the power connector from the back of it. > In my case it is as easy to do a DHCP release and renew > > I see this problem with customers. A firmware update to > the router may help. > Doesn't seem to be available. > In addition, this from Linksys: > > Based on customer feedback, North America E-mail support will > be discontinued starting May 15th, 2007. E-mail received > before that date will be handled and responses will be sent > out until June 1st, when the email system will be shut down > completely. > > It could be the router itself. > Only seems to happen when CON cast places me in a specific IP > range with a short lease time!! > > Other brands of consumer grade routers do this. I rarely see this > > behavior with more expensive routers like those from sonic. > > > I have a used SonicWall router here, but have yet to see if I > can set it up I also have a computer with Monowall installed, > but couldn't figure out how to open up the port for CNET. > I SUPPOSE I could also use my Vonage V1005V. > Or simply tough it through until CONCAST stops playing games. > > JN > > > Lee > > > > -----Original Message----- > > From: voip-bounces at ckts.info > [mailto:voip-bounces at ckts.info] On Behalf > > Of John Novack > > Sent: Friday, May 11, 2007 10:25 PM > > To: Voice Over IP > > Subject: [VoIP] CON cast > > > > CON cast is at it again. They seem to be on a mission to > change my IP > > address and a short DHCP lease time, then my Linksys router > fails to > > ask for a renewal, so all goes dead until I goose it from > the setup screens. > > They have gone through this before, then all settles down > for a while. > > I suppose their IT boys are up to something, but it sure is a pain. > > If you find my CNET connection down at a given time, that > is probably > > the reason > > > > Wonder if I went to another brand of router things would improve? > > > > 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/ > > _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From firecomm at tecnj.org Sat May 12 23:14:38 2007 From: firecomm at tecnj.org (FireComm) Date: Sun, 13 May 2007 00:14:38 -0400 Subject: [VoIP] Smart 1 for cost of shipping In-Reply-To: <026801c794f8$11247270$336d5750$@com> Message-ID: <20070513001437.017dad25@clnts50net02.clftn.tecnj.org> I have a Smart 1 we found in the stock room, it is missing the power supply, but its up for grabs for the cost of shipping, anyone interested email me directly with your address. T The information contained in this E-mail message is privileged, confidential, and may be protected from disclosure; please be aware that any other use, printing, copying, publishing, disclosure or dissemination of this communication may be subject to legal restriction or sanction. If you think that you have received this E-mail message in error, please reply to the sender immediately. This E-mail message and any attachments have been scanned for viruses and are believed to be free of any virus or other defect that might affect any computer system into which it is received and opened. However, it is the responsibility of the recipient to ensure that it is virus free and no responsibility is accepted by the Tri-State Emergency Communications Consortium, Its Member Organizations, Subsidiaries or Affiliates for any loss or damage arising in any way from its use. From firecomm at tecnj.org Sat May 12 23:18:31 2007 From: firecomm at tecnj.org (FireComm) Date: Sun, 13 May 2007 00:18:31 -0400 Subject: [VoIP] Smart 1 for the cost of shipping In-Reply-To: <026801c794f8$11247270$336d5750$@com> Message-ID: <20070513001831.97c2721c@clnts50net02.clftn.tecnj.org> I have a Smart 1 we found in the stock room last week, it is missing the power supply, but its up for grabs for the cost of shipping, anyone interested email me directly with your address. P The information contained in this E-mail message is privileged, confidential, and may be protected from disclosure; please be aware that any other use, printing, copying, publishing, disclosure or dissemination of this communication may be subject to legal restriction or sanction. If you think that you have received this E-mail message in error, please reply to the sender immediately. This E-mail message and any attachments have been scanned for viruses and are believed to be free of any virus or other defect that might affect any computer system into which it is received and opened. However, it is the responsibility of the recipient to ensure that it is virus free and no responsibility is accepted by the Tri-State Emergency Communications Consortium, Its Member Organizations, Subsidiaries or Affiliates for any loss or damage arising in any way from its use. From jnovack at stromberg-carlson.org Tue May 15 20:11:35 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Tue, 15 May 2007 21:11:35 -0400 Subject: [VoIP] Valid CLID Message-ID: <464A5A47.70506@stromberg-carlson.org> Who is this?? 347 Dimitri Ressetar I really wish that callers would send some sort of reasonable call back number, rather than 3470000 which terminates nowhere Just one of my personal pet peeves. I suppose the GOOD news is that with all of CONCASTs fooling around, ANYONE can reach me!! John Novack From jnovack at stromberg-carlson.org Tue May 15 20:12:07 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Tue, 15 May 2007 21:12:07 -0400 Subject: [VoIP] Additional information Message-ID: <464A5A67.6050604@stromberg-carlson.org> 2007-05-15 19:08:06 22 cnetIAX 13470000 6669969 extensions ""Harrisburg PA" <13470000> IAX2/72.70.168.96:50935-4 Hangup ANSWERED From lee at spenadel.com Tue May 15 20:27:22 2007 From: lee at spenadel.com (Lee Spenadel) Date: Tue, 15 May 2007 21:27:22 -0400 Subject: [VoIP] Valid CLID In-Reply-To: <464A5A47.70506@stromberg-carlson.org> References: <464A5A47.70506@stromberg-carlson.org> Message-ID: <00af01c79759$5a56ccf0$0f0466d0$@com> What? You're not feeling the love? -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Tuesday, May 15, 2007 9:12 PM To: Voice Over IP Subject: [VoIP] Valid CLID Who is this?? 347 Dimitri Ressetar I really wish that callers would send some sort of reasonable call back number, rather than 3470000 which terminates nowhere Just one of my personal pet peeves. I suppose the GOOD news is that with all of CONCASTs fooling around, ANYONE can reach me!! John Novack _______________________________________________ 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 Tue May 15 20:43:05 2007 From: david at josephson.com (David Josephson) Date: Tue, 15 May 2007 18:43:05 -0700 Subject: [VoIP] Valid CLID In-Reply-To: <464A5A47.70506@stromberg-carlson.org> References: <464A5A47.70506@stromberg-carlson.org> Message-ID: <464A61A9.3070503@josephson.com> John Novack wrote: > Who is this?? > > 347 Dimitri Ressetar > > I really wish that callers would send some sort of reasonable call back number, rather than 3470000 which terminates nowhere > > Just one of my personal pet peeves. We're all working on getting CLID defined correctly as calls go through our switches. Imagine my surprise when I got a call on my prepaid Liechtenstein GSM cell phone when I was in Austria last week, showing the extension number in my internal numbering system and nothing else, when one of the folks here called me. Some VoIP carriers accept and pass on whatever digits they get. -- David Josephson From mitya89 at aim.com Tue May 15 22:06:53 2007 From: mitya89 at aim.com (Dimitri Ressetar) Date: Tue, 15 May 2007 23:06:53 -0400 Subject: [VoIP] Valid CLID In-Reply-To: References: Message-ID: <464A754D.40203@aim.com> Mr. Novack: I'm sorry I haven't posted on the mailing list yet nor got my caller id working quite right. I contacted Greg Blakely back in Nov 06 about connecting to the network, and I've been experimenting with it and Linux/Asterisk servers on and off as I have had time (I am a high school student). Right now, I don't have my server running all the time because I'm still configuring it and writing my dialplan. It is rarely on longer than a few hours at a time, so I haven't posted to the list yet nor added numbers to the phone book. I set the outgoing CID to 3470000 to at least make sure my calls were routeable back to my switch, and hopefully I'll get individual extensions CID to work over the network soon. Sorry about that! (I'll also have to check my dialplan since I'm writing it by hand and route 347000 to something sensible for the time being.) I have finals at school this week, but in the next few weeks I will have some more free time and hopefully I can get the server running all the time and then I'll make another post to the list. -Dimitri Ressetar dimitri at ressetar dot com cnet 347-9454 (when my server is on) ---- Subject: [VoIP] Valid CLID From: John Novack Date: Tue, 15 May 2007 21:11:35 -0400 To: Voice Over IP Who is this?? 347 Dimitri Ressetar I really wish that callers would send some sort of reasonable call back number, rather than 3470000 which terminates nowhere Just one of my personal pet peeves. I suppose the GOOD news is that with all of CONCASTs fooling around, ANYONE can reach me!! John Novack ------------------------------------------------------------------------ Subject: [VoIP] Additional information From: John Novack Date: Tue, 15 May 2007 21:12:07 -0400 To: Voice Over IP 2007-05-15 19:08:06 22 cnetIAX 13470000 6669969 extensions ""Harrisburg PA" <13470000> IAX2/72.70.168.96:50935-4 Hangup ANSWERED From hockd at dteenergy.com Wed May 16 07:02:52 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Wed, 16 May 2007 08:02:52 -0400 Subject: [VoIP] Valid CLID Message-ID: Dmitri, First Welcome to CNet! John is a great resource as are manyu pof the others you will find on the CNet. I myself am Office Code 269 (spells COW, don't ask, long story). It does help if you can send a valid number. However, It is great that you are so young and working on this. I am 54 and having CRS and brain farts on this newer technology. I work at a SE Mich power utlity as a Principal Engineer in the Network Engineering - Telephony group. We have a 42 switch voice network. Big thing to remember is this. It is a learning , living , lab envirnoment. It is also envisioned as a safe haven where it is ok to ask questions, make mistakes and try things. Great to have you on board. What year of high school are you in? and if you care to where are you located? Have you seen all the foreign places we already reach in the short time we have been in existence. Best to you, Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: voip at ckts.info, jnovack at stromberg-carlson.org From: Dimitri Ressetar Sent by: voip-bounces at ckts.info Date: 05/15/2007 11:06PM Subject: Re: [VoIP] Valid CLID Mr. Novack: I'm sorry I haven't posted on the mailing list yet nor got my caller id working quite right. I contacted Greg Blakely back in Nov 06 about connecting to the network, and I've been experimenting with it and Linux/Asterisk servers on and off as I have had time (I am a high school student). Right now, I don't have my server running all the time because I'm still configuring it and writing my dialplan. It is rarely on longer than a few hours at a time, so I haven't posted to the list yet nor added numbers to the phone book. I set the outgoing CID to 3470000 to at least make sure my calls were routeable back to my switch, and hopefully I'll get individual extensions CID to work over the network soon. Sorry about that! (I'll also have to check my dialplan since I'm writing it by hand and route 347000 to something sensible for the time being.) I have finals at school this week, but in the next few weeks I will have some more free time and hopefully I can get the server running all the time and then I'll make another post to the list. -Dimitri Ressetar dimitri at ressetar dot com cnet 347-9454 (when my server is on) ---- Subject: [VoIP] Valid CLID From: John Novack Date: Tue, 15 May 2007 21:11:35 -0400 To: Voice Over IP Who is this?? 347 Dimitri Ressetar I really wish that callers would send some sort of reasonable call back number, rather than 3470000 which terminates nowhere Just one of my personal pet peeves. I suppose the GOOD news is that with all of CONCASTs fooling around, ANYONE can reach me!! John Novack ------------------------------------------------------------------------ Subject: [VoIP] Additional information From: John Novack Date: Tue, 15 May 2007 21:12:07 -0400 To: Voice Over IP 2007-05-15 19:08:06 22 cnetIAX 13470000 6669969 extensions ""Harrisburg PA" <13470000> IAX2/72.70.168.96:50935-4 Hangup ANSWERED _______________________________________________ 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 Wed May 16 09:45:24 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Wed, 16 May 2007 10:45:24 -0400 Subject: [VoIP] Valid CLID In-Reply-To: <464A754D.40203@aim.com> References: <464A754D.40203@aim.com> Message-ID: <464B1904.5080409@stromberg-carlson.org> Welcome to CNET as well. My box is on all the time, so feel free to give it or me a call whenever. Most of the numbers are in the CNET directory, though I am having some EM switch problems at the moment and not enough time to track them down I can usually be reached at 6669900. I do hope MOST of my CLID numbers are correct. There is another caller who only sends 2 digits "33" with a name of GT&T. I am more than happy to help anyone work through problems on their dialplan, but there are quite a few more skilled and knowledgeable than I in this area. I can handle simple things. for the more difficult John C or Shane Young, as well as Lee S . John Novack Dimitri Ressetar wrote: > Mr. Novack: > > I'm sorry I haven't posted on the mailing list yet nor got my caller id > working quite right. I contacted Greg Blakely back in Nov 06 about > connecting to the network, and I've been experimenting with it and > Linux/Asterisk servers on and off as I have had time (I am a high school > student). Right now, I don't have my server running all the time because > I'm still configuring it and writing my dialplan. It is rarely on longer > than a few hours at a time, so I haven't posted to the list yet nor > added numbers to the phone book. I set the outgoing CID to 3470000 to at > least make sure my calls were routeable back to my switch, and hopefully > I'll get individual extensions CID to work over the network soon. Sorry > about that! (I'll also have to check my dialplan since I'm writing it by > hand and route 347000 to something sensible for the time being.) I have > finals at school this week, but in the next few weeks I will have some > more free time and hopefully I can get the server running all the time > and then I'll make another post to the list. > > -Dimitri Ressetar > dimitri at ressetar dot com > cnet 347-9454 (when my server is on) > > ---- > Subject: > [VoIP] Valid CLID > From: > John Novack > Date: > Tue, 15 May 2007 21:11:35 -0400 > > To: > Voice Over IP > > > Who is this?? > > 347 Dimitri Ressetar > > I really wish that callers would send some sort of reasonable call back > number, rather than 3470000 which terminates nowhere > > Just one of my personal pet peeves. > > > > > I suppose the GOOD news is that with all of CONCASTs fooling around, > ANYONE can reach me!! > John Novack > > > > > > > > > ------------------------------------------------------------------------ > > Subject: > [VoIP] Additional information > From: > John Novack > Date: > Tue, 15 May 2007 21:12:07 -0400 > > To: > Voice Over IP > > > 2007-05-15 19:08:06 22 cnetIAX 13470000 6669969 > extensions ""Harrisburg PA" <13470000> > IAX2/72.70.168.96:50935-4 > Hangup ANSWERED > > From voiptandem at shaneyoung.com Wed May 16 11:09:25 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Wed, 16 May 2007 11:09:25 -0500 Subject: [VoIP] Valid CLID In-Reply-To: <464B1904.5080409@stromberg-carlson.org> References: <464A754D.40203@aim.com> <464B1904.5080409@stromberg-carlson.org> Message-ID: <20070516110925.o594bqwjqo8cgwkw@mail.shaneyoung.com> So, this brings up a good topic for discussion... What *should* we use as a valid CLID? Here are some examples of what I've received: VT1005_Line_1 9550000 2184886566 12639900 +12691212 I'm pretty sure when I send CLID it's 10 digits with 200 being the NPA like 2008217383 I *think* that 7 digits in the US is what most people are sending. Quoting John Novack : > Welcome to CNET as well. > > My box is on all the time, so feel free to give it or me a call > whenever. Most of the numbers are in the CNET directory, though I am > having some EM switch problems at the moment and not enough time to > track them down > I can usually be reached at 6669900. > I do hope MOST of my CLID numbers are correct. > There is another caller who only sends 2 digits "33" with a name of GT&T. > I am more than happy to help anyone work through problems on their > dialplan, but there are quite a few more skilled and knowledgeable than > I in this area. > I can handle simple things. for the more difficult John C or Shane > Young, as well as Lee S . > > > John Novack > Dimitri Ressetar wrote: >> Mr. Novack: >> >> I'm sorry I haven't posted on the mailing list yet nor got my caller id >> working quite right. I contacted Greg Blakely back in Nov 06 about >> connecting to the network, and I've been experimenting with it and >> Linux/Asterisk servers on and off as I have had time (I am a high school >> student). Right now, I don't have my server running all the time because >> I'm still configuring it and writing my dialplan. It is rarely on longer >> than a few hours at a time, so I haven't posted to the list yet nor >> added numbers to the phone book. I set the outgoing CID to 3470000 to at >> least make sure my calls were routeable back to my switch, and hopefully >> I'll get individual extensions CID to work over the network soon. Sorry >> about that! (I'll also have to check my dialplan since I'm writing it by >> hand and route 347000 to something sensible for the time being.) I have >> finals at school this week, but in the next few weeks I will have some >> more free time and hopefully I can get the server running all the time >> and then I'll make another post to the list. >> >> -Dimitri Ressetar >> dimitri at ressetar dot com >> cnet 347-9454 (when my server is on) >> >> ---- >> Subject: >> [VoIP] Valid CLID >> From: >> John Novack >> Date: >> Tue, 15 May 2007 21:11:35 -0400 >> >> To: >> Voice Over IP >> >> >> Who is this?? >> >> 347 Dimitri Ressetar >> >> I really wish that callers would send some sort of reasonable call back >> number, rather than 3470000 which terminates nowhere >> >> Just one of my personal pet peeves. >> >> >> >> >> I suppose the GOOD news is that with all of CONCASTs fooling around, >> ANYONE can reach me!! >> John Novack >> >> >> >> >> >> >> >> >> ------------------------------------------------------------------------ >> >> Subject: >> [VoIP] Additional information >> From: >> John Novack >> Date: >> Tue, 15 May 2007 21:12:07 -0400 >> >> To: >> Voice Over IP >> >> >> 2007-05-15 19:08:06 22 cnetIAX 13470000 6669969 >> extensions ""Harrisburg PA" <13470000> >> IAX2/72.70.168.96:50935-4 >> Hangup ANSWERED >> >> > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > --Shane +1-821-7311 CNET From hockd at dteenergy.com Wed May 16 11:10:34 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Wed, 16 May 2007 12:10:34 -0400 Subject: [VoIP] Valid CLID Message-ID: John you are being too modest, not to detract from anyone else. ;-) Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Dimitri Ressetar From: John Novack Sent by: voip-bounces at ckts.info Date: 05/16/2007 10:45AM cc: voip at ckts.info Subject: Re: [VoIP] Valid CLID Welcome to CNET as well. My box is on all the time, so feel free to give it or me a call whenever. Most of the numbers are in the CNET directory, though I am having some EM switch problems at the moment and not enough time to track them down I can usually be reached at 6669900. I do hope MOST of my CLID numbers are correct. There is another caller who only sends 2 digits "33" with a name of GT&T. I am more than happy to help anyone work through problems on their dialplan, but there are quite a few more skilled and knowledgeable than I in this area. I can handle simple things. for the more difficult John C or Shane Young, as well as Lee S . John Novack Dimitri Ressetar wrote: > Mr. Novack: > > I'm sorry I haven't posted on the mailing list yet nor got my caller id > working quite right. I contacted Greg Blakely back in Nov 06 about > connecting to the network, and I've been experimenting with it and > Linux/Asterisk servers on and off as I have had time (I am a high school > student). Right now, I don't have my server running all the time because > I'm still configuring it and writing my dialplan. It is rarely on longer > than a few hours at a time, so I haven't posted to the list yet nor > added numbers to the phone book. I set the outgoing CID to 3470000 to at > least make sure my calls were routeable back to my switch, and hopefully > I'll get individual extensions CID to work over the network soon. Sorry > about that! (I'll also have to check my dialplan since I'm writing it by > hand and route 347000 to something sensible for the time being.) I have > finals at school this week, but in the next few weeks I will have some > more free time and hopefully I can get the server running all the time > and then I'll make another post to the list. > > -Dimitri Ressetar > dimitri at ressetar dot com > cnet 347-9454 (when my server is on) > > ---- > Subject: > [VoIP] Valid CLID > From: > John Novack > Date: > Tue, 15 May 2007 21:11:35 -0400 > > To: > Voice Over IP > > > Who is this?? > > 347 Dimitri Ressetar > > I really wish that callers would send some sort of reasonable call back > number, rather than 3470000 which terminates nowhere > > Just one of my personal pet peeves. > > > > > I suppose the GOOD news is that with all of CONCASTs fooling around, > ANYONE can reach me!! > John Novack > > > > > > > > > ------------------------------------------------------------------------ > > Subject: > [VoIP] Additional information > From: > John Novack > Date: > Tue, 15 May 2007 21:12:07 -0400 > > To: > Voice Over IP > > > 2007-05-15 19:08:06 22 cnetIAX 13470000 6669969 > extensions ""Harrisburg PA" <13470000> > IAX2/72.70.168.96:50935-4 > Hangup ANSWERED > > _______________________________________________ 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 Wed May 16 14:43:34 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Wed, 16 May 2007 15:43:34 -0400 Subject: [VoIP] Valid CLID In-Reply-To: <20070516110925.o594bqwjqo8cgwkw@mail.shaneyoung.com> References: <464A754D.40203@aim.com> <464B1904.5080409@stromberg-carlson.org> <20070516110925.o594bqwjqo8cgwkw@mail.shaneyoung.com> Message-ID: <464B5EE6.5020302@stromberg-carlson.org> Shane Young wrote: > So, this brings up a good topic for discussion... > > What *should* we use as a valid CLID? > Since I opened my mouth, I would suggest sending a valid CNET number that could be returned. Perhaps some even have the ability, or should would, do a "return call" so either 1+NXX-XXXX or NXX-XXXX for North America. Sorry my mind STILL can't get around the open numbering plan to offer a suggestion there. Sending the "200" NPA never made much sense to me either. Perhaps I can't see where CNET will ever have to go beyond 7 digit dialing for North America, and the current International scheme. > Here are some examples of what I've received: > > VT1005_Line_1 > Fairly simple to change the Motorola to use a 7 digit number. I have found that the user name and number must match, which seems to be an Asterisk limitation. I could be wrong on that, as it has been a couple of months since I messed with the VT 1005 > 9550000 > Simple enough to have each device report its CLID, or use one CLID that reflects a good callback number. Also would be nice to either send a name or some device identifier. I probably need to re examine some of mine as well for that. > 2184886566 > Guessing that is someones PSTN number? > 12639900 > That should also work, if the dialplan is programmed correctly. I allow dialing as either 7 digits, 1 plus seven digits for North America, though I never dial it that way myself my portal will accept that format > +12691212 > Bet that is Dennis Hock through Greg's switch. I suppose a return call could parse that into a good CNET number > I'm pretty sure when I send CLID it's 10 digits with 200 being the NPA > like 2008217383 > Why not simply get rid of the 200 NPA?? > I *think* that 7 digits in the US is what most people are sending. > CNET expects 1 plus 7 digits for the DNS/ENUM to work properly, but 7 digit CLID for North America is what I consider ideal. Just my opinion, though John Novack From hockd at dteenergy.com Mon May 21 09:16:44 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Mon, 21 May 2007 10:16:44 -0400 Subject: [VoIP] Valid CLID Message-ID: I agree with John. Iknow we have had this discussion previously I guess I haven't had my mind on it enough to understand but I don't see the need for the 200 yet, if ever in CNET. I think if for the US members or NANP members we send either a 1nxx-xxxx or nxx-xxxx as the CLID for our calls with the caveat that in general it be a dialable somehow answered number even if only a announcement of some type. As a thought if the day should come do we want / need to use a NPA should we immediately apply a 3 digit NPA or possibly simply break the NANPA into say ten location areas? Each area could be a single digit allowing a good section of the country to be encompassed under a single digit NPA code. Although this veers away from current practice it might make it more palatible for many of us who would like to dial a standard set of yet minimum digits. Thoughts anyone? That +12691212 is my number throguh Gregs tandem. Not exactly sure why the + is appended onto the front of the number. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: John Novack Sent by: voip-bounces at ckts.info Date: 05/16/2007 03:43PM Subject: Re: [VoIP] Valid CLID Shane Young wrote: > So, this brings up a good topic for discussion... > > What *should* we use as a valid CLID? > Since I opened my mouth, I would suggest sending a valid CNET number that could be returned. Perhaps some even have the ability, or should would, do a "return call" so either 1+NXX-XXXX or NXX-XXXX for North America. Sorry my mind STILL can't get around the open numbering plan to offer a suggestion there. Sending the "200" NPA never made much sense to me either. Perhaps I can't see where CNET will ever have to go beyond 7 digit dialing for North America, and the current International scheme. > Here are some examples of what I've received: > > VT1005_Line_1 > Fairly simple to change the Motorola to use a 7 digit number. I have found that the user name and number must match, which seems to be an Asterisk limitation. I could be wrong on that, as it has been a couple of months since I messed with the VT 1005 > 9550000 > Simple enough to have each device report its CLID, or use one CLID that reflects a good callback number. Also would be nice to either send a name or some device identifier. I probably need to re examine some of mine as well for that. > 2184886566 > Guessing that is someones PSTN number? > 12639900 > That should also work, if the dialplan is programmed correctly. I allow dialing as either 7 digits, 1 plus seven digits for North America, though I never dial it that way myself my portal will accept that format > +12691212 > Bet that is Dennis Hock through Greg's switch. I suppose a return call could parse that into a good CNET number > I'm pretty sure when I send CLID it's 10 digits with 200 being the NPA > like 2008217383 > Why not simply get rid of the 200 NPA?? > I *think* that 7 digits in the US is what most people are sending. > CNET expects 1 plus 7 digits for the DNS/ENUM to work properly, but 7 digit CLID for North America is what I consider ideal. Just my opinion, though John Novack _______________________________________________ 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 Mon May 21 10:30:40 2007 From: voiptandem at shaneyoung.com (Shane Young) Date: Mon, 21 May 2007 10:30:40 -0500 Subject: [VoIP] Valid CLID In-Reply-To: References: Message-ID: <20070521103040.sca8f3tyfk8sgw4c@mail.shaneyoung.com> The + indicates it's supposed to be in e.164 format which includes the country code (1 in this case) My preference would be to send non-NA ANI as +country-code and the number North ameican could be sent as either 7 digits or +1 and seven digits. Quoting Dennis D Hock : > > I agree with John. Iknow we have had this discussion previously I guess I > haven't had my mind on it enough to understand but I don't see the need for > the 200 yet, if ever in CNET. I think if for the US members or NANP > members we send either a 1nxx-xxxx or nxx-xxxx as the CLID for our calls > with the caveat that in general it be a dialable somehow answered number > even if only a announcement of some type. > > As a thought if the day should come do we want / need to use a NPA should > we immediately apply a 3 digit NPA or possibly simply break the NANPA into > say ten location areas? Each area could be a single digit allowing a good > section of the country to be encompassed under a single digit NPA code. > Although this veers away from current practice it might make it more > palatible for many of us who would like to dial a standard set of yet > minimum digits. Thoughts anyone? > > That +12691212 is my number throguh Gregs tandem. Not exactly sure why > the + is appended onto the front of the number. > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: John Novack > Sent by: voip-bounces at ckts.info > Date: 05/16/2007 03:43PM > Subject: Re: [VoIP] Valid CLID > > > > Shane Young wrote: >> So, this brings up a good topic for discussion... >> >> What *should* we use as a valid CLID? >> > Since I opened my mouth, I would suggest sending a valid CNET number > that could be returned. Perhaps some even have the ability, or should > would, do a "return call" so either 1+NXX-XXXX or NXX-XXXX for North > America. > Sorry my mind STILL can't get around the open numbering plan to offer a > suggestion there. > Sending the "200" NPA never made much sense to me either. Perhaps I > can't see where CNET will ever have to go beyond 7 digit dialing for > North America, and the current International scheme. > >> Here are some examples of what I've received: >> >> VT1005_Line_1 >> > Fairly simple to change the Motorola to use a 7 digit number. I have > found that the user name and number must match, which seems to be an > Asterisk limitation. I could be wrong on that, as it has been a couple > of months since I messed with the VT 1005 >> 9550000 >> > Simple enough to have each device report its CLID, or use one CLID that > reflects a good callback number. > Also would be nice to either send a name or some device identifier. I > probably need to re examine some of mine as well for that. >> 2184886566 >> > Guessing that is someones PSTN number? >> 12639900 >> > That should also work, if the dialplan is programmed correctly. > I allow dialing as either 7 digits, 1 plus seven digits for North > America, though I never dial it that way myself my portal will accept > that format >> +12691212 >> > Bet that is Dennis Hock through Greg's switch. I suppose a return call > could parse that into a good CNET number >> I'm pretty sure when I send CLID it's 10 digits with 200 being the NPA >> like 2008217383 >> > Why not simply get rid of the 200 NPA?? > >> I *think* that 7 digits in the US is what most people are sending. >> > CNET expects 1 plus 7 digits for the DNS/ENUM to work properly, but 7 > digit CLID for North America is what I consider ideal. > > > Just my opinion, though > > John Novack > > _______________________________________________ > 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/ > --Shane +1-821-7311 CNET From hockd at dteenergy.com Mon May 21 14:25:00 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Mon, 21 May 2007 15:25:00 -0400 Subject: [VoIP] Valid CLID Message-ID: Opps first part was the brain fart. The second part sounds good to me but I think maybe there should be more input from some of the others. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: voip at ckts.info From: Shane Young Sent by: voip-bounces at ckts.info Date: 05/21/2007 11:30AM Subject: Re: [VoIP] Valid CLID The + indicates it's supposed to be in e.164 format which includes the country code (1 in this case) My preference would be to send non-NA ANI as +country-code and the number North ameican could be sent as either 7 digits or +1 and seven digits. Quoting Dennis D Hock : > > I agree with John. Iknow we have had this discussion previously I guess I > haven't had my mind on it enough to understand but I don't see the need for > the 200 yet, if ever in CNET. I think if for the US members or NANP > members we send either a 1nxx-xxxx or nxx-xxxx as the CLID for our calls > with the caveat that in general it be a dialable somehow answered number > even if only a announcement of some type. > > As a thought if the day should come do we want / need to use a NPA should > we immediately apply a 3 digit NPA or possibly simply break the NANPA into > say ten location areas? Each area could be a single digit allowing a good > section of the country to be encompassed under a single digit NPA code. > Although this veers away from current practice it might make it more > palatible for many of us who would like to dial a standard set of yet > minimum digits. Thoughts anyone? > > That +12691212 is my number throguh Gregs tandem. Not exactly sure why > the + is appended onto the front of the number. > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: John Novack > Sent by: voip-bounces at ckts.info > Date: 05/16/2007 03:43PM > Subject: Re: [VoIP] Valid CLID > > > > Shane Young wrote: >> So, this brings up a good topic for discussion... >> >> What *should* we use as a valid CLID? >> > Since I opened my mouth, I would suggest sending a valid CNET number > that could be returned. Perhaps some even have the ability, or should > would, do a "return call" so either 1+NXX-XXXX or NXX-XXXX for North > America. > Sorry my mind STILL can't get around the open numbering plan to offer a > suggestion there. > Sending the "200" NPA never made much sense to me either. Perhaps I > can't see where CNET will ever have to go beyond 7 digit dialing for > North America, and the current International scheme. > >> Here are some examples of what I've received: >> >> VT1005_Line_1 >> > Fairly simple to change the Motorola to use a 7 digit number. I have > found that the user name and number must match, which seems to be an > Asterisk limitation. I could be wrong on that, as it has been a couple > of months since I messed with the VT 1005 >> 9550000 >> > Simple enough to have each device report its CLID, or use one CLID that > reflects a good callback number. > Also would be nice to either send a name or some device identifier. I > probably need to re examine some of mine as well for that. >> 2184886566 >> > Guessing that is someones PSTN number? >> 12639900 >> > That should also work, if the dialplan is programmed correctly. > I allow dialing as either 7 digits, 1 plus seven digits for North > America, though I never dial it that way myself my portal will accept > that format >> +12691212 >> > Bet that is Dennis Hock through Greg's switch. I suppose a return call > could parse that into a good CNET number >> I'm pretty sure when I send CLID it's 10 digits with 200 being the NPA >> like 2008217383 >> > Why not simply get rid of the 200 NPA?? > >> I *think* that 7 digits in the US is what most people are sending. >> > CNET expects 1 plus 7 digits for the DNS/ENUM to work properly, but 7 > digit CLID for North America is what I consider ideal. > > > Just my opinion, though > > John Novack > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voipProjectWeb Page: > http://www.ckts.info/ > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > --Shane +1-821-7311 CNET _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From greg at vyger.net Fri May 25 22:40:55 2007 From: greg at vyger.net (Greg Blakely) Date: Fri, 25 May 2007 22:40:55 -0500 Subject: [VoIP] Vonage vs Verizon Message-ID: It looks like Vonage is finally trying to assert the "obviousness" of the ip-to-pstn lookup technology. But it may be a day late and a dollar short. Verizon is claiming that the "obviousness" case that Vonage is citing in its appeal was already decided by the time that Vonage and Verizon were in court, and that, if Vonage wanted to assert that the patent should never have been granted because the procedure was obvious to those in the know of those sorts of things, they should have done it at trial, and failed to do so. Whether Vonage pulls out of this or not, it is apparent that the obviousness of ENUM lookups will make it tough for Verizon to go after other voip providers. The full story is at http://www.internetnews.com/bus-news/article.php/3679581. From mark at rudholm.com Sat May 26 23:14:15 2007 From: mark at rudholm.com (Mark Rudholm) Date: Sat, 26 May 2007 21:14:15 -0700 Subject: [VoIP] Numbering Message-ID: <46590597.1070204@rudholm.com> Greetings, I'm not presently a member (don't have any vintage switchgear) but know of cnet through a friend who is a participant. I run a couple of Asterisk installations, and am thinking about numbering options, particularly for interconnecting multiple Asterisks. I'm looking at various ways to direct calls off a switch, such as fake Feature Group D dialing prefixes, or fake NPAs, or simply assigning portions of internal numberspace to a given external switch. In any case, I figured I'd read through the mailing list archives here to see how cnet has solved the problem. For some reason, I was under the impression that cnet was using 311 as an NPA, but I guess that isn't the case and individual members must simply be integrating cnet numberspace into their numbering plans in various ways they devise personally. Or just aren't integrating PSTN and cnet numberspace. How are folks doing this? Or are they? For cnet, it seems like using an nonassignable "NPA" like 311 would make sense, since it would let you seamlessly integrate cnet numbers into the PSTN dialing without collision problems. The mentions of ENUM also piqued my interest. Is cnet running an ENUM DNS server somewhere? It'd be neat if it was actually under 1.1.3.1.e164.arpa. or some such. -Mark From stfkerman at jps.net Sat May 26 23:27:36 2007 From: stfkerman at jps.net (Steph Kerman) Date: Sun, 27 May 2007 00:27:36 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <46590597.1070204@rudholm.com> References: <46590597.1070204@rudholm.com> Message-ID: <465908B8.3010800@jps.net> Actually, 311 is being used in some places within the NANP. NYC uses it for non-emergency calls to municipal agencies to get these calls off 911. ISTR talk about something similar being proposed for a city in the SF Bay Area a few years ago, perhaps San Jose. Steph Mark Rudholm wrote: > Greetings, I'm not presently a member (don't have any vintage switchgear) > but know of cnet through a friend who is a participant. > > I run a couple of Asterisk installations, and am thinking about numbering > options, particularly for interconnecting multiple Asterisks. I'm looking > at various ways to direct calls off a switch, such as fake Feature Group D > dialing prefixes, or fake NPAs, or simply assigning portions of internal > numberspace to a given external switch. > > In any case, I figured I'd read through the mailing list archives here to > see how cnet has solved the problem. For some reason, I was under the > impression that cnet was using 311 as an NPA, but I guess that isn't the > case and individual members must simply be integrating cnet numberspace > into their numbering plans in various ways they devise personally. Or > just aren't integrating PSTN and cnet numberspace. > > How are folks doing this? Or are they? For cnet, it seems like using > an nonassignable "NPA" like 311 would make sense, since it would let you > seamlessly integrate cnet numbers into the PSTN dialing without > collision problems. > > The mentions of ENUM also piqued my interest. Is cnet running an ENUM > DNS server somewhere? It'd be neat if it was actually under > 1.1.3.1.e164.arpa. or some such. > > -Mark > > http://www.ckts.info/ > > From mark at rudholm.com Sat May 26 23:41:57 2007 From: mark at rudholm.com (Mark Rudholm) Date: Sat, 26 May 2007 21:41:57 -0700 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <465908B8.3010800@jps.net> References: <46590597.1070204@rudholm.com> <465908B8.3010800@jps.net> Message-ID: <46590C15.4090302@rudholm.com> Steph Kerman wrote: > Actually, 311 is being used in some places within the NANP. NYC uses > it for non-emergency calls to municipal agencies to get these calls off > 911. ISTR talk about something similar being proposed for a city in the > SF Bay Area a few years ago, perhaps San Jose. That makes "311" less attractive, since it couldn't be used in a 10D dialing scheme, but it still could be used in 1+10D dialing. But regardless of the "NPA" used, is this question not on the table? Are people integrating numbering? If so, how? -Mark From greg at vyger.net Sun May 27 10:00:35 2007 From: greg at vyger.net (Greg Blakely) Date: Sun, 27 May 2007 10:00:35 -0500 Subject: [VoIP] Numbering - existing use of 311 Message-ID: I have been dual-assigning member entries -- one with NPA 200, and one without. So, from my perspective, NPA 200 would be the easiest to implement, since it already is implemented. What would be needed is for the individual switchers to set the asterisk boxes to accept either 1200NXXXXXX or 1NXXXXXX. I believe that some of us already are setup for that. > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of Mark Rudholm > Sent: Saturday, May 26, 2007 11:42 PM > To: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Numbering - existing use of 311 > > Steph Kerman wrote: > > Actually, 311 is being used in some places within the > NANP. NYC uses > > it for non-emergency calls to municipal agencies to get these calls > > off 911. ISTR talk about something similar being proposed > for a city > > in the SF Bay Area a few years ago, perhaps San Jose. > > That makes "311" less attractive, since it couldn't be used > in a 10D dialing scheme, but it still could be used in 1+10D dialing. > > But regardless of the "NPA" used, is this question not on the > table? Are people integrating numbering? If so, how? > > -Mark > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From mark at rudholm.com Sun May 27 10:15:00 2007 From: mark at rudholm.com (Mark Rudholm) Date: Sun, 27 May 2007 08:15:00 -0700 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: References: Message-ID: <4659A074.6060803@rudholm.com> Greg Blakely wrote: > I have been dual-assigning member entries -- one with NPA 200, and one > without. > > So, from my perspective, NPA 200 would be the easiest to implement, > since it already is implemented. > > What would be needed is for the individual switchers to set the asterisk > boxes to accept either 1200NXXXXXX or 1NXXXXXX. > > I believe that some of us already are setup for that. OK, that's easy enough. Although I'd probably leave out the 1NXXXXXX since that collides with regular 1NXXNXXXXXX dialing, making a timeout necessary. Is cnet using an ENUM DNS server? From jnovack at stromberg-carlson.org Sun May 27 15:02:59 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Sun, 27 May 2007 16:02:59 -0400 Subject: [VoIP] Numbering In-Reply-To: <46590597.1070204@rudholm.com> References: <46590597.1070204@rudholm.com> Message-ID: <4659E3F3.8010000@stromberg-carlson.org> Mark Rudholm wrote: > Greetings, I'm not presently a member (don't have any vintage switchgear)but know of cnet through a friend who is a participant. > And you don't necessarily NEED to have vintage gear, simply the hope and desire to someday have some. Some members don't currently have any, some simply some Key equipment and at the other end at least one fellow out West has over 1000 lines and several different technologies working or near working. Not sure where in the world you might be, but in the US, at least, there still is a quantity of gear around and available. You need : 1. The desire 2. The space 3. the willingness and ability and time to go after it. Warning: It is a slippery slope. Once you have some REAL switching gear, you will soon discover how primitive Asterisk really is in certain areas. Unfortunately, the original design was done with out the benefit of some knowledge of history and the current state of the art. Slowly, ever so slowly, some of those shortcomings are being addressed. There was a group of users that were using 311, but when Greg redid things after an invasion, they seemed to have disappeared. Currently there are a couple of free portals from the PSTN into CNET, but given the current costs of "real" PSTN numbers and the fact that most of us don't have really deep pockets, inward calling to CNET is done the way mobile roaming was done in the early 80's Some of us have PSTN outward calling via prepaid SIP services that USUALLY works well. I have yet to find an IAX service though. Within our private network things work pretty well, give the differences in skill level of the members. Welcome to the group and hope you can participate. John Novack CNET 666-9900 From mark at rudholm.com Sun May 27 16:08:04 2007 From: mark at rudholm.com (Mark Rudholm) Date: Sun, 27 May 2007 14:08:04 -0700 Subject: [VoIP] Numbering In-Reply-To: <4659E3F3.8010000@stromberg-carlson.org> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> Message-ID: <4659F334.1020906@rudholm.com> John Novack wrote: > > Mark Rudholm wrote: >> Greetings, I'm not presently a member (don't have any vintage switchgear)but know of cnet through a friend who is a participant. >> > And you don't necessarily NEED to have vintage gear, simply the hope and > desire to someday have some. Some members don't currently have any, some > simply some Key equipment and at the other end at least one fellow out > West has over 1000 lines and several different technologies working or > near working. I suspect the fellow of whom you speak is the same person as the friend of whom I spoke. :-) > Not sure where in the world you might be, but in the US, at least, there > still is a quantity of gear around and available. I'm in Los Angeles (District Area 5, to be specific) > You need : > 1. The desire > 2. The space > 3. the willingness and ability and time to go after it. My hurdle would probably be the space, as I'd have to allocate some garage space for it. Plus, I pay up to 27 cents per kWh, so I'm averse to any significant constant electrical loads. Having a proper switch would be nice, though, since, amongst other things, it'd enable me to avoid having to graft Coin Service functionality onto Asterisk (I don't have any proper switch gear, but I do have a number of proper payphones that I'm currently running off a telco DMS100 17Q Coin Service line). Hmm, what's the smallest, not necessarily vintage, real switch that could provide DTF coin service and wouldn't eat a lot of electricity? > Warning: It is a slippery slope. Once you have some REAL switching gear, > you will soon discover how primitive Asterisk really is in certain I'm aware. > areas. Unfortunately, the original design was done with out the benefit > of some knowledge of history and the current state of the art. Slowly, Yeah, Mark readily acknowledges this. I was talking to him about coin service signaling the other day. He didn't have a telephone background when he created Asterisk and I don't think he knew how popular it would eventually become. > ever so slowly, some of those shortcomings are being addressed. > There was a group of users that were using 311, but when Greg redid > things after an invasion, they seemed to have disappeared. > Currently there are a couple of free portals from the PSTN into CNET, > but given the current costs of "real" PSTN numbers and the fact that > most of us don't have really deep pockets, inward calling to CNET is > done the way mobile roaming was done in the early 80's I already have a route into aforementioned fellow's Asterisk, so I could use that as a CNET gateway as well. > Some of us have PSTN outward calling via prepaid SIP services that > USUALLY works well. > I have yet to find an IAX service though. VoipJet is cheap. Teliax is good. Both offer call completion via IAX2 (Teliax also offers PSTN numbers for inbound). > Within our private network things work pretty well, give the > differences in skill level of the members. > > Welcome to the group and hope you can participate. Thanks. Sounds like there's a lot of knowledgeable folk here. I like that. -Mark From lee at spenadel.com Sun May 27 17:15:31 2007 From: lee at spenadel.com (Lee Spenadel) Date: Sun, 27 May 2007 18:15:31 -0400 Subject: [VoIP] Numbering In-Reply-To: <4659F334.1020906@rudholm.com> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> <4659F334.1020906@rudholm.com> Message-ID: <011a01c7a0ac$8b39ab60$a1ad0220$@com> Mark, I built my own Step-by-Step switch. It lives in a standard telco data rack (7' high) and has 3 line finders, 4 connectors (one is a Coin Connector for my payphones), 3 selectors and two trunk cans. I interface this to CNET via Asterisk - office code 349 in the CNET directory. I installed an Adtran 850 channel bank and T1 card in the * box. Works nicely. * became the center of my telephony universe here, allowing me to tie together multiple systems. The best part of my SxS is the support for my 10 3-slot payphones, all of which are configured for Coin First service - with answer supervision. Get on board. Reserve an office code. You'll be surprised at what you start to build....... Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of Mark Rudholm Sent: Sunday, May 27, 2007 5:08 PM To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering John Novack wrote: > > Mark Rudholm wrote: >> Greetings, I'm not presently a member (don't have any vintage switchgear)but know of cnet through a friend who is a participant. >> > And you don't necessarily NEED to have vintage gear, simply the hope and > desire to someday have some. Some members don't currently have any, some > simply some Key equipment and at the other end at least one fellow out > West has over 1000 lines and several different technologies working or > near working. I suspect the fellow of whom you speak is the same person as the friend of whom I spoke. :-) > Not sure where in the world you might be, but in the US, at least, there > still is a quantity of gear around and available. I'm in Los Angeles (District Area 5, to be specific) > You need : > 1. The desire > 2. The space > 3. the willingness and ability and time to go after it. My hurdle would probably be the space, as I'd have to allocate some garage space for it. Plus, I pay up to 27 cents per kWh, so I'm averse to any significant constant electrical loads. Having a proper switch would be nice, though, since, amongst other things, it'd enable me to avoid having to graft Coin Service functionality onto Asterisk (I don't have any proper switch gear, but I do have a number of proper payphones that I'm currently running off a telco DMS100 17Q Coin Service line). Hmm, what's the smallest, not necessarily vintage, real switch that could provide DTF coin service and wouldn't eat a lot of electricity? > Warning: It is a slippery slope. Once you have some REAL switching gear, > you will soon discover how primitive Asterisk really is in certain I'm aware. > areas. Unfortunately, the original design was done with out the benefit > of some knowledge of history and the current state of the art. Slowly, Yeah, Mark readily acknowledges this. I was talking to him about coin service signaling the other day. He didn't have a telephone background when he created Asterisk and I don't think he knew how popular it would eventually become. > ever so slowly, some of those shortcomings are being addressed. > There was a group of users that were using 311, but when Greg redid > things after an invasion, they seemed to have disappeared. > Currently there are a couple of free portals from the PSTN into CNET, > but given the current costs of "real" PSTN numbers and the fact that > most of us don't have really deep pockets, inward calling to CNET is > done the way mobile roaming was done in the early 80's I already have a route into aforementioned fellow's Asterisk, so I could use that as a CNET gateway as well. > Some of us have PSTN outward calling via prepaid SIP services that > USUALLY works well. > I have yet to find an IAX service though. VoipJet is cheap. Teliax is good. Both offer call completion via IAX2 (Teliax also offers PSTN numbers for inbound). > Within our private network things work pretty well, give the > differences in skill level of the members. > > Welcome to the group and hope you can participate. Thanks. Sounds like there's a lot of knowledgeable folk here. I like that. -Mark _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From mark at rudholm.com Sun May 27 18:28:19 2007 From: mark at rudholm.com (Mark Rudholm) Date: Sun, 27 May 2007 16:28:19 -0700 Subject: [VoIP] Numbering In-Reply-To: <011a01c7a0ac$8b39ab60$a1ad0220$@com> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> <4659F334.1020906@rudholm.com> <011a01c7a0ac$8b39ab60$a1ad0220$@com> Message-ID: <465A1413.5050301@rudholm.com> Lee Spenadel wrote: > Mark, > > I built my own Step-by-Step switch. It lives in a standard telco data rack > (7' high) and has 3 line finders, 4 connectors (one is a Coin Connector for > my payphones), 3 selectors and two trunk cans. I interface this to CNET via > Asterisk - office code 349 in the CNET directory. I installed an Adtran 850 > channel bank and T1 card in the * box. Works nicely. * became the center > of my telephony universe here, allowing me to tie together multiple systems. > The best part of my SxS is the support for my 10 3-slot payphones, all of > which are configured for Coin First service - with answer supervision. > > Get on board. Reserve an office code. You'll be surprised at what you > start to build....... Hmm... it would be nice to have the 3-slot in my kitchen working right. Although, I'm not sure where I'd get the operator for calls requiring more than the initial rate deposit :-) I don't suppose anyone has a complete set of ACTS recordings I might have a copy of? Speaking of ACTS, it's a shame it no longer supports inter-LATA. The other night, just out of curiosity, I dialed some 0+ inter-LATA calls from the 2C2 in my study and the nice lady told me that no, there's no more coin paid service for long distance anymore and that I'd have to use a credit/calling card (at quite ridiculous prices). Now that AT&T is back in the payphone business (by virtue of merging with RBOCs that are in the payphone business) they really ought to resume inter-LATA service. From jjones3601 at yahoo.com Sun May 27 19:15:46 2007 From: jjones3601 at yahoo.com (john jones) Date: Sun, 27 May 2007 17:15:46 -0700 (PDT) Subject: [VoIP] Numbering Message-ID: <520036.54860.qm@web34307.mail.mud.yahoo.com> Lee, Do you have any information you can share on your coin setup? The number of the coin connector, additional support equipment, etc? Thanks! John ----- Original Message ---- From: Lee Spenadel To: Voice Over IP Tandem for Analog Switches Sent: Sunday, May 27, 2007 6:15:31 PM Subject: Re: [VoIP] Numbering Mark, I built my own Step-by-Step switch. It lives in a standard telco data rack (7' high) and has 3 line finders, 4 connectors (one is a Coin Connector for my payphones), 3 selectors and two trunk cans. I interface this to CNET via Asterisk - office code 349 in the CNET directory. I installed an Adtran 850 channel bank and T1 card in the * box. Works nicely. * became the center of my telephony universe here, allowing me to tie together multiple systems. The best part of my SxS is the support for my 10 3-slot payphones, all of which are configured for Coin First service - with answer supervision. Get on board. Reserve an office code. You'll be surprised at what you start to build....... Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of Mark Rudholm Sent: Sunday, May 27, 2007 5:08 PM To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering John Novack wrote: > > Mark Rudholm wrote: >> Greetings, I'm not presently a member (don't have any vintage switchgear)but know of cnet through a friend who is a participant. >> > And you don't necessarily NEED to have vintage gear, simply the hope and > desire to someday have some. Some members don't currently have any, some > simply some Key equipment and at the other end at least one fellow out > West has over 1000 lines and several different technologies working or > near working. I suspect the fellow of whom you speak is the same person as the friend of whom I spoke. :-) > Not sure where in the world you might be, but in the US, at least, there > still is a quantity of gear around and available. I'm in Los Angeles (District Area 5, to be specific) > You need : > 1. The desire > 2. The space > 3. the willingness and ability and time to go after it. My hurdle would probably be the space, as I'd have to allocate some garage space for it. Plus, I pay up to 27 cents per kWh, so I'm averse to any significant constant electrical loads. Having a proper switch would be nice, though, since, amongst other things, it'd enable me to avoid having to graft Coin Service functionality onto Asterisk (I don't have any proper switch gear, but I do have a number of proper payphones that I'm currently running off a telco DMS100 17Q Coin Service line). Hmm, what's the smallest, not necessarily vintage, real switch that could provide DTF coin service and wouldn't eat a lot of electricity? > Warning: It is a slippery slope. Once you have some REAL switching gear, > you will soon discover how primitive Asterisk really is in certain I'm aware. > areas. Unfortunately, the original design was done with out the benefit > of some knowledge of history and the current state of the art. Slowly, Yeah, Mark readily acknowledges this. I was talking to him about coin service signaling the other day. He didn't have a telephone background when he created Asterisk and I don't think he knew how popular it would eventually become. > ever so slowly, some of those shortcomings are being addressed. > There was a group of users that were using 311, but when Greg redid > things after an invasion, they seemed to have disappeared. > Currently there are a couple of free portals from the PSTN into CNET, > but given the current costs of "real" PSTN numbers and the fact that > most of us don't have really deep pockets, inward calling to CNET is > done the way mobile roaming was done in the early 80's I already have a route into aforementioned fellow's Asterisk, so I could use that as a CNET gateway as well. > Some of us have PSTN outward calling via prepaid SIP services that > USUALLY works well. > I have yet to find an IAX service though. VoipJet is cheap. Teliax is good. Both offer call completion via IAX2 (Teliax also offers PSTN numbers for inbound). > Within our private network things work pretty well, give the > differences in skill level of the members. > > Welcome to the group and hope you can participate. Thanks. Sounds like there's a lot of knowledgeable folk here. I like that. -Mark _______________________________________________ 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 Sun May 27 19:35:24 2007 From: stfkerman at jps.net (Steph Kerman) Date: Sun, 27 May 2007 20:35:24 -0400 Subject: [VoIP] Numbering In-Reply-To: <465A1413.5050301@rudholm.com> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> <4659F334.1020906@rudholm.com> <011a01c7a0ac$8b39ab60$a1ad0220$@com> <465A1413.5050301@rudholm.com> Message-ID: <465A23CC.7030702@jps.net> Is the 2C2 connected to a coin line or a POTS line? Steph Mark Rudholm wrote: > Speaking of ACTS, it's a shame it no longer supports inter-LATA. > The other night, just out of curiosity, I dialed some 0+ inter-LATA > calls from the 2C2 in my study and the nice lady told me that no, > there's no more coin paid service for long distance anymore and > that I'd have to use a credit/calling card (at quite ridiculous > prices). Now that AT&T is back in the payphone business (by virtue > of merging with RBOCs that are in the payphone business) they really > ought to resume inter-LATA service. > > From stfkerman at jps.net Sun May 27 19:43:04 2007 From: stfkerman at jps.net (Steph Kerman) Date: Sun, 27 May 2007 20:43:04 -0400 Subject: [VoIP] Numbering - cost of power In-Reply-To: <4659F334.1020906@rudholm.com> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> <4659F334.1020906@rudholm.com> Message-ID: <465A2598.5090307@jps.net> Electromechanical switches often consume very little or actually no power at all at 48V when idle. Any continuous power consumption is usually the standby loss in the 120VAC to 48V DC power conversion . If you find an efficient switcher, that can be very low too. Avoid ferro-resonant supplies, which are expensive to idle. Besides the acoustic noise, even a linear supply can be more efficient because if the load duty cycle is low, as it usually is in a hobby exchange, the power loss from heat dissipation in the linear regulation can be more than compensated by low standby consumption. Of course if there were heavy traffic, quite the opposite would be true. But keeping a line frequency transformer powered is likely to cost more than keeping a switcher powered. Steph Mark Rudholm wrote: > Plus, I pay up to 27 cents per kWh, so I'm averse to any significant constant electrical loads. > > Hmm, what's the smallest, not necessarily vintage, real switch that could provide DTF coin service and wouldn't eat a lot of electricity? > From mark at rudholm.com Sun May 27 20:57:20 2007 From: mark at rudholm.com (Mark Rudholm) Date: Sun, 27 May 2007 18:57:20 -0700 Subject: [VoIP] Numbering In-Reply-To: <465A23CC.7030702@jps.net> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> <4659F334.1020906@rudholm.com> <011a01c7a0ac$8b39ab60$a1ad0220$@com> <465A1413.5050301@rudholm.com> <465A23CC.7030702@jps.net> Message-ID: <465A3700.70406@rudholm.com> Steph Kerman wrote: > Is the 2C2 connected to a coin line or a POTS line? It's connected to a 17Q Coin Service line off a DMS100. From stfkerman at jps.net Sun May 27 21:01:58 2007 From: stfkerman at jps.net (Steph Kerman) Date: Sun, 27 May 2007 22:01:58 -0400 Subject: [VoIP] Numbering - 2C2 CTS In-Reply-To: <465A3700.70406@rudholm.com> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> <4659F334.1020906@rudholm.com> <011a01c7a0ac$8b39ab60$a1ad0220$@com> <465A1413.5050301@rudholm.com> <465A23CC.7030702@jps.net> <465A3700.70406@rudholm.com> Message-ID: <465A3816.10004@jps.net> That's interesting! Semi-public service? What do you have to pay for that? Steph Mark Rudholm wrote: > Steph Kerman wrote: > >> Is the 2C2 connected to a coin line or a POTS line? >> > > It's connected to a 17Q Coin Service line off a DMS100. > _______________________________________________ > 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 May 27 21:15:40 2007 From: lee at spenadel.com (Lee Spenadel) Date: Sun, 27 May 2007 22:15:40 -0400 Subject: [VoIP] Numbering In-Reply-To: <520036.54860.qm@web34307.mail.mud.yahoo.com> References: <520036.54860.qm@web34307.mail.mud.yahoo.com> Message-ID: <012d01c7a0ce$170fff60$452ffe20$@com> John, The coin connector is an AE: DH61876 B40A / 8H61876 B1 A-WK492. The only support equipment is a 130V power supply that the coin connector uses to send the proper collect/return signal to the coin relay based on whether the call was answered or not (supervision). Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of john jones Sent: Sunday, May 27, 2007 8:16 PM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering Lee, Do you have any information you can share on your coin setup? The number of the coin connector, additional support equipment, etc? Thanks! John ----- Original Message ---- From: Lee Spenadel To: Voice Over IP Tandem for Analog Switches Sent: Sunday, May 27, 2007 6:15:31 PM Subject: Re: [VoIP] Numbering Mark, I built my own Step-by-Step switch. It lives in a standard telco data rack (7' high) and has 3 line finders, 4 connectors (one is a Coin Connector for my payphones), 3 selectors and two trunk cans. I interface this to CNET via Asterisk - office code 349 in the CNET directory. I installed an Adtran 850 channel bank and T1 card in the * box. Works nicely. * became the center of my telephony universe here, allowing me to tie together multiple systems. The best part of my SxS is the support for my 10 3-slot payphones, all of which are configured for Coin First service - with answer supervision. Get on board. Reserve an office code. You'll be surprised at what you start to build....... Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of Mark Rudholm Sent: Sunday, May 27, 2007 5:08 PM To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering John Novack wrote: > > Mark Rudholm wrote: >> Greetings, I'm not presently a member (don't have any vintage switchgear)but know of cnet through a friend who is a participant. >> > And you don't necessarily NEED to have vintage gear, simply the hope and > desire to someday have some. Some members don't currently have any, some > simply some Key equipment and at the other end at least one fellow out > West has over 1000 lines and several different technologies working or > near working. I suspect the fellow of whom you speak is the same person as the friend of whom I spoke. :-) > Not sure where in the world you might be, but in the US, at least, there > still is a quantity of gear around and available. I'm in Los Angeles (District Area 5, to be specific) > You need : > 1. The desire > 2. The space > 3. the willingness and ability and time to go after it. My hurdle would probably be the space, as I'd have to allocate some garage space for it. Plus, I pay up to 27 cents per kWh, so I'm averse to any significant constant electrical loads. Having a proper switch would be nice, though, since, amongst other things, it'd enable me to avoid having to graft Coin Service functionality onto Asterisk (I don't have any proper switch gear, but I do have a number of proper payphones that I'm currently running off a telco DMS100 17Q Coin Service line). Hmm, what's the smallest, not necessarily vintage, real switch that could provide DTF coin service and wouldn't eat a lot of electricity? > Warning: It is a slippery slope. Once you have some REAL switching gear, > you will soon discover how primitive Asterisk really is in certain I'm aware. > areas. Unfortunately, the original design was done with out the benefit > of some knowledge of history and the current state of the art. Slowly, Yeah, Mark readily acknowledges this. I was talking to him about coin service signaling the other day. He didn't have a telephone background when he created Asterisk and I don't think he knew how popular it would eventually become. > ever so slowly, some of those shortcomings are being addressed. > There was a group of users that were using 311, but when Greg redid > things after an invasion, they seemed to have disappeared. > Currently there are a couple of free portals from the PSTN into CNET, > but given the current costs of "real" PSTN numbers and the fact that > most of us don't have really deep pockets, inward calling to CNET is > done the way mobile roaming was done in the early 80's I already have a route into aforementioned fellow's Asterisk, so I could use that as a CNET gateway as well. > Some of us have PSTN outward calling via prepaid SIP services that > USUALLY works well. > I have yet to find an IAX service though. VoipJet is cheap. Teliax is good. Both offer call completion via IAX2 (Teliax also offers PSTN numbers for inbound). > Within our private network things work pretty well, give the > differences in skill level of the members. > > Welcome to the group and hope you can participate. Thanks. Sounds like there's a lot of knowledgeable folk here. I like that. -Mark _______________________________________________ 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 Sun May 27 21:32:31 2007 From: jjones3601 at yahoo.com (john jones) Date: Sun, 27 May 2007 19:32:31 -0700 (PDT) Subject: [VoIP] Numbering Message-ID: <742446.72103.qm@web34315.mail.mud.yahoo.com> Lee, Thanks for the information. Do you have any pdf files of the DH61876? John ----- Original Message ---- From: Lee Spenadel To: Voice Over IP Tandem for Analog Switches Sent: Sunday, May 27, 2007 10:15:40 PM Subject: Re: [VoIP] Numbering John, The coin connector is an AE: DH61876 B40A / 8H61876 B1 A-WK492. The only support equipment is a 130V power supply that the coin connector uses to send the proper collect/return signal to the coin relay based on whether the call was answered or not (supervision). Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of john jones Sent: Sunday, May 27, 2007 8:16 PM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering Lee, Do you have any information you can share on your coin setup? The number of the coin connector, additional support equipment, etc? Thanks! John ----- Original Message ---- From: Lee Spenadel To: Voice Over IP Tandem for Analog Switches Sent: Sunday, May 27, 2007 6:15:31 PM Subject: Re: [VoIP] Numbering Mark, I built my own Step-by-Step switch. It lives in a standard telco data rack (7' high) and has 3 line finders, 4 connectors (one is a Coin Connector for my payphones), 3 selectors and two trunk cans. I interface this to CNET via Asterisk - office code 349 in the CNET directory. I installed an Adtran 850 channel bank and T1 card in the * box. Works nicely. * became the center of my telephony universe here, allowing me to tie together multiple systems. The best part of my SxS is the support for my 10 3-slot payphones, all of which are configured for Coin First service - with answer supervision. Get on board. Reserve an office code. You'll be surprised at what you start to build....... Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of Mark Rudholm Sent: Sunday, May 27, 2007 5:08 PM To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering John Novack wrote: > > Mark Rudholm wrote: >> Greetings, I'm not presently a member (don't have any vintage switchgear)but know of cnet through a friend who is a participant. >> > And you don't necessarily NEED to have vintage gear, simply the hope and > desire to someday have some. Some members don't currently have any, some > simply some Key equipment and at the other end at least one fellow out > West has over 1000 lines and several different technologies working or > near working. I suspect the fellow of whom you speak is the same person as the friend of whom I spoke. :-) > Not sure where in the world you might be, but in the US, at least, there > still is a quantity of gear around and available. I'm in Los Angeles (District Area 5, to be specific) > You need : > 1. The desire > 2. The space > 3. the willingness and ability and time to go after it. My hurdle would probably be the space, as I'd have to allocate some garage space for it. Plus, I pay up to 27 cents per kWh, so I'm averse to any significant constant electrical loads. Having a proper switch would be nice, though, since, amongst other things, it'd enable me to avoid having to graft Coin Service functionality onto Asterisk (I don't have any proper switch gear, but I do have a number of proper payphones that I'm currently running off a telco DMS100 17Q Coin Service line). Hmm, what's the smallest, not necessarily vintage, real switch that could provide DTF coin service and wouldn't eat a lot of electricity? > Warning: It is a slippery slope. Once you have some REAL switching gear, > you will soon discover how primitive Asterisk really is in certain I'm aware. > areas. Unfortunately, the original design was done with out the benefit > of some knowledge of history and the current state of the art. Slowly, Yeah, Mark readily acknowledges this. I was talking to him about coin service signaling the other day. He didn't have a telephone background when he created Asterisk and I don't think he knew how popular it would eventually become. > ever so slowly, some of those shortcomings are being addressed. > There was a group of users that were using 311, but when Greg redid > things after an invasion, they seemed to have disappeared. > Currently there are a couple of free portals from the PSTN into CNET, > but given the current costs of "real" PSTN numbers and the fact that > most of us don't have really deep pockets, inward calling to CNET is > done the way mobile roaming was done in the early 80's I already have a route into aforementioned fellow's Asterisk, so I could use that as a CNET gateway as well. > Some of us have PSTN outward calling via prepaid SIP services that > USUALLY works well. > I have yet to find an IAX service though. VoipJet is cheap. Teliax is good. Both offer call completion via IAX2 (Teliax also offers PSTN numbers for inbound). > Within our private network things work pretty well, give the > differences in skill level of the members. > > Welcome to the group and hope you can participate. Thanks. Sounds like there's a lot of knowledgeable folk here. I like that. -Mark _______________________________________________ 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 mark at rudholm.com Sun May 27 21:56:42 2007 From: mark at rudholm.com (Mark Rudholm) Date: Sun, 27 May 2007 19:56:42 -0700 Subject: [VoIP] Numbering - 2C2 CTS In-Reply-To: <465A3816.10004@jps.net> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> <4659F334.1020906@rudholm.com> <011a01c7a0ac$8b39ab60$a1ad0220$@com> <465A1413.5050301@rudholm.com> <465A23CC.7030702@jps.net> <465A3700.70406@rudholm.com> <465A3816.10004@jps.net> Message-ID: <465A44EA.3050809@rudholm.com> Steph Kerman wrote: > That's interesting! Semi-public service? What do you have to pay for that? The service is provided by SBC/AT&T. The intended market is Payphone Service Providers (i.e. COCOT operators) who want to use real payphones using telco-side billing/control (ACTS) rather than internal logic. I use it simply because I have a couple 2C2s (one is new-old stock!) a 1D2, and a 2D2. It's kind of expensive, about 65$/month after taxes. No, I haven't tried putting more than one payphone on it at a time. At worst, I suspect it wouldn't work at all (since the coin tests wouldn't work right) and at best, it'd be odd, since the coin relay in all connected phones would actuate in unison. Oh, it's not a 17Q, it's a 1PC (the 17Q is, electrically, no different from a regular business phone line, it's the 1PC that has the coin signaling). The 17Q is what they delivered at first, in error. Dunno who your telco is, but for SBC/AT&T, go here: https://primeaccess.att.com/ (and click on "Payphone") That site is interesting. You can buy pretty much anything these days from AT&T if you're willing to pay for it. Such a change from the bad old days when the telcos were fortresses and you couldn't so much as order Touch-Tone service without proving that you hadn't stolen one of their Touch-Tone phones. From stfkerman at jps.net Sun May 27 23:38:43 2007 From: stfkerman at jps.net (Steph Kerman) Date: Mon, 28 May 2007 00:38:43 -0400 Subject: [VoIP] Numbering - 2C2 CTS In-Reply-To: <465A44EA.3050809@rudholm.com> References: <46590597.1070204@rudholm.com> <4659E3F3.8010000@stromberg-carlson.org> <4659F334.1020906@rudholm.com> <011a01c7a0ac$8b39ab60$a1ad0220$@com> <465A1413.5050301@rudholm.com> <465A23CC.7030702@jps.net> <465A3700.70406@rudholm.com> <465A3816.10004@jps.net> <465A44EA.3050809@rudholm.com> Message-ID: <465A5CD3.6000302@jps.net> Mark Rudholm wrote: > Steph Kerman wrote: >> That's interesting! Semi-public service? What do you have to pay for >> that? > The service is provided by SBC/AT&T. The intended market is Payphone > Service Providers (i.e. COCOT operators) who want to use real > payphones using telco-side billing/control (ACTS) rather than internal > logic. I use it simply because I have a couple 2C2s (one is new-old > stock!) a 1D2, and a 2D2. > > It's kind of expensive, about 65$/month after taxes. > > No, I haven't tried putting more than one payphone on it at a time. At > worst, I suspect it wouldn't work at all (since the coin tests > wouldn't work right) and at best, it'd be odd, since the coin relay in > all connected phones would actuate in unison. That certainly would not be a problem with 3-slot phones on a CF line. The coin relay is disconnected unless there is a coin in the hopper. I believe the same is true with single slot phones except that there is a ground isolation circuit to prevent ground noise. A single slot phone provides 2 indications back to the CO: any coin present and initial rate present. I suspect that phones with no coin on deposit would be completely passive to both tests as well as coin disposal attempts. But there might be some problem with the Totalizer. I just don't know. So if I were you, I would try it. There's nothing to lose. In any case, you could always insert a simple relay concentrator between the coin phones and line so that only the first phone that attempts to seize the line is connected to it and the others are locked out. Then you can put one of your less pristine looking 1D2s on a pedestal out by the street and earn some revenue to cover some of that $65/month expense! Until your town zoning code enforcement people show up... There is an arrangement to provide a manual answering extension on coin lines terminated in single slot CTSs. The answering telephone contains a special circuit board containing a mercury wetted relay and a bunch of simple semiconductor devices. I think one purpose of the circuit in the manual answering phone is to prevent it from interfering with outgoing calls to avoid loss of deposited coins. The BSP does not contain a schematic of the circuit board and although I do have a sample, I not reverse engineered it... yet. It might provide some insight into the issues involved in coin line sharing though. > Oh, it's not a 17Q, it's a 1PC (the 17Q is, electrically, no different > from a regular business phone line, it's the 1PC > that has the coin signaling). The 17Q is what they delivered at first, > in error. > > Dunno who your telco is, but for SBC/AT&T, go here: > https://primeaccess.att.com/ (and click on "Payphone") Thanks > That site is interesting. You can buy pretty much anything these days > from AT&T if you're willing to pay for it. Such a change from the bad > old days when the telcos were fortresses and you couldn't so much as > order Touch-Tone service without proving that you hadn't stolen one of > their Touch-Tone phones. Yes. A different world. Better in some ways and worse in others. At the moment, the only coin phone I have set up, a 3-slot 2-piece 150G, is handled by a simple prepay adapter circuit I built a few decades ago. (How time flies!) The prepay adapter circuit is installed between the phone and the line circuit of my crossbar PBX. It makes the phone function as a nickel prepay PBX extension with 9th level access to the PSTN. Steph From lee at spenadel.com Mon May 28 21:21:27 2007 From: lee at spenadel.com (Lee Spenadel) Date: Mon, 28 May 2007 22:21:27 -0400 Subject: [VoIP] Numbering In-Reply-To: <742446.72103.qm@web34315.mail.mud.yahoo.com> References: <742446.72103.qm@web34315.mail.mud.yahoo.com> Message-ID: <000e01c7a198$105e87c0$311b9740$@com> John, I only have it in its original size. Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of john jones Sent: Sunday, May 27, 2007 10:33 PM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering Lee, Thanks for the information. Do you have any pdf files of the DH61876? John ----- Original Message ---- From: Lee Spenadel To: Voice Over IP Tandem for Analog Switches Sent: Sunday, May 27, 2007 10:15:40 PM Subject: Re: [VoIP] Numbering John, The coin connector is an AE: DH61876 B40A / 8H61876 B1 A-WK492. The only support equipment is a 130V power supply that the coin connector uses to send the proper collect/return signal to the coin relay based on whether the call was answered or not (supervision). Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of john jones Sent: Sunday, May 27, 2007 8:16 PM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering Lee, Do you have any information you can share on your coin setup? The number of the coin connector, additional support equipment, etc? Thanks! John ----- Original Message ---- From: Lee Spenadel To: Voice Over IP Tandem for Analog Switches Sent: Sunday, May 27, 2007 6:15:31 PM Subject: Re: [VoIP] Numbering Mark, I built my own Step-by-Step switch. It lives in a standard telco data rack (7' high) and has 3 line finders, 4 connectors (one is a Coin Connector for my payphones), 3 selectors and two trunk cans. I interface this to CNET via Asterisk - office code 349 in the CNET directory. I installed an Adtran 850 channel bank and T1 card in the * box. Works nicely. * became the center of my telephony universe here, allowing me to tie together multiple systems. The best part of my SxS is the support for my 10 3-slot payphones, all of which are configured for Coin First service - with answer supervision. Get on board. Reserve an office code. You'll be surprised at what you start to build....... Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of Mark Rudholm Sent: Sunday, May 27, 2007 5:08 PM To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Numbering John Novack wrote: > > Mark Rudholm wrote: >> Greetings, I'm not presently a member (don't have any vintage switchgear)but know of cnet through a friend who is a participant. >> > And you don't necessarily NEED to have vintage gear, simply the hope and > desire to someday have some. Some members don't currently have any, some > simply some Key equipment and at the other end at least one fellow out > West has over 1000 lines and several different technologies working or > near working. I suspect the fellow of whom you speak is the same person as the friend of whom I spoke. :-) > Not sure where in the world you might be, but in the US, at least, there > still is a quantity of gear around and available. I'm in Los Angeles (District Area 5, to be specific) > You need : > 1. The desire > 2. The space > 3. the willingness and ability and time to go after it. My hurdle would probably be the space, as I'd have to allocate some garage space for it. Plus, I pay up to 27 cents per kWh, so I'm averse to any significant constant electrical loads. Having a proper switch would be nice, though, since, amongst other things, it'd enable me to avoid having to graft Coin Service functionality onto Asterisk (I don't have any proper switch gear, but I do have a number of proper payphones that I'm currently running off a telco DMS100 17Q Coin Service line). Hmm, what's the smallest, not necessarily vintage, real switch that could provide DTF coin service and wouldn't eat a lot of electricity? > Warning: It is a slippery slope. Once you have some REAL switching gear, > you will soon discover how primitive Asterisk really is in certain I'm aware. > areas. Unfortunately, the original design was done with out the benefit > of some knowledge of history and the current state of the art. Slowly, Yeah, Mark readily acknowledges this. I was talking to him about coin service signaling the other day. He didn't have a telephone background when he created Asterisk and I don't think he knew how popular it would eventually become. > ever so slowly, some of those shortcomings are being addressed. > There was a group of users that were using 311, but when Greg redid > things after an invasion, they seemed to have disappeared. > Currently there are a couple of free portals from the PSTN into CNET, > but given the current costs of "real" PSTN numbers and the fact that > most of us don't have really deep pockets, inward calling to CNET is > done the way mobile roaming was done in the early 80's I already have a route into aforementioned fellow's Asterisk, so I could use that as a CNET gateway as well. > Some of us have PSTN outward calling via prepaid SIP services that > USUALLY works well. > I have yet to find an IAX service though. VoipJet is cheap. Teliax is good. Both offer call completion via IAX2 (Teliax also offers PSTN numbers for inbound). > Within our private network things work pretty well, give the > differences in skill level of the members. > > Welcome to the group and hope you can participate. Thanks. Sounds like there's a lot of knowledgeable folk here. I like that. -Mark _______________________________________________ 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 hockd at dteenergy.com Wed May 30 05:25:56 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Wed, 30 May 2007 06:25:56 -0400 Subject: [VoIP] Numbering - existing use of 311 Message-ID: Yes 311 has been designated for non emergency access to ones supposed local govt unit. I think the confusion here is in calling 311 a NPA. I believe the Notes on the Network indicate that it and all n11 codes are really SAC and have no real NPA designation. To add to what John and lee and Stephg and others have said Welcome. You don't need to have any old equipment per see, just the thirst and curiosity. Thanks to several folks here I am on line and continuing to rebuild my Step by Step (Strowger) switch a snails pace but there has been some progress. This is a great group of people. They are all willing to share and help and this is a wonderful safe place to learn and ask questions without fear of reprisal. I have to agree with what one of the responders said sign up for an Office Code and join in. Take care, Dennis H. -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: Steph Kerman Sent by: voip-bounces at ckts.info Date: 05/27/2007 12:27AM Subject: Re: [VoIP] Numbering - existing use of 311 Actually, 311 is being used in some places within the NANP. NYC uses it for non-emergency calls to municipal agencies to get these calls off 911. ISTR talk about something similar being proposed for a city in the SF Bay Area a few years ago, perhaps San Jose. Steph Mark Rudholm wrote: > Greetings, I'm not presently a member (don't have any vintage switchgear) > but know of cnet through a friend who is a participant. > > I run a couple of Asterisk installations, and am thinking about numbering > options, particularly for interconnecting multiple Asterisks. I'm looking > at various ways to direct calls off a switch, such as fake Feature Group D > dialing prefixes, or fake NPAs, or simply assigning portions of internal > numberspace to a given external switch. > > In any case, I figured I'd read through the mailing list archives here to > see how cnet has solved the problem. For some reason, I was under the > impression that cnet was using 311 as an NPA, but I guess that isn't the > case and individual members must simply be integrating cnet numberspace > into their numbering plans in various ways they devise personally. Or > just aren't integrating PSTN and cnet numberspace. > > How are folks doing this? Or are they? For cnet, it seems like using > an nonassignable "NPA" like 311 would make sense, since it would let you > seamlessly integrate cnet numbers into the PSTN dialing without > collision problems. > > The mentions of ENUM also piqued my interest. Is cnet running an ENUM > DNS server somewhere? It'd be neat if it was actually under > 1.1.3.1.e164.arpa. or some such. > > -Mark > > 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 stfkerman at jps.net Wed May 30 07:18:48 2007 From: stfkerman at jps.net (Steph Kerman) Date: Wed, 30 May 2007 08:18:48 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: References: Message-ID: <465D6BA8.2000207@jps.net> Aren't there areas of the country where 10-digit dialing is used without 1+? Or are do these only allow local or intra-LATA dialing to those few nearby NPAs? Steph Dennis D Hock wrote: > Yes 311 has been designated for non emergency access to ones supposed local > govt unit. I think the confusion here is in calling 311 a NPA. I believe > the Notes on the Network indicate that it and all n11 codes are really SAC > and have no real NPA designation. > > T > From jnovack at stromberg-carlson.org Wed May 30 08:41:30 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Wed, 30 May 2007 09:41:30 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <465D6BA8.2000207@jps.net> References: <465D6BA8.2000207@jps.net> Message-ID: <465D7F0A.6080303@stromberg-carlson.org> The whole nationwide dialing situation is a mess. Some areas have 10 digit LOCAL dialing, 1 plus for toll dialing, others require 11 digit for local AND toll, and others conform to neither. Here in Berkeley county WV, Verizon has 7 digit local INTERstate dialing to MD 301, BUT, if you need to call the overlay 240, you need to dial 11 digits. The MD part of the LATA requires 10 digit for ALL local calls, and 1 plus for toll calls. Let's leave CNET as 7 (8) digit dialing until we NEED more office codes in North America. Anyone is, of course, free to structure their internal dialplan as they choose. I have mine set up to accept 7 digits, add the 1 for NA and use 011 + CC + number for others It works well for me. John Novack Steph Kerman wrote: > Aren't there areas of the country where 10-digit dialing is used without > 1+? Or are do these only allow local or intra-LATA dialing to those few > nearby NPAs? > > Steph > > Dennis D Hock wrote: > >> Yes 311 has been designated for non emergency access to ones supposed local >> govt unit. I think the confusion here is in calling 311 a NPA. I believe >> the Notes on the Network indicate that it and all n11 codes are really SAC >> and have no real NPA designation. >> >> T >> >> > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From Googol at ciudad.com.ar Wed May 30 12:32:28 2007 From: Googol at ciudad.com.ar (Googol@ciudad.com.ar) Date: Wed, 30 May 2007 14:32:28 -0300 Subject: [VoIP] =?iso-8859-1?q?_VT1005_config_tools?= Message-ID: <20070530143228.A1A18523.B9EE29B4@172.16.1.38> hello I am of Argentina, and I have bought a motorola vt1005s, but there are no local suppliers of voip that know to form via east TFTP device, so that here it is not used systems of provisioning, looking for, I read east article of John Novack, where is the file of provisioning, but encounter the program bticonfig.exe not to compile it, if somebody could give me to this program or another program with its respective configuration file and a brief explanation, would be to me of much utility and very it would be been thankful him. used traslator spanish - english From david at josephson.com Wed May 30 14:28:36 2007 From: david at josephson.com (David Josephson) Date: Wed, 30 May 2007 12:28:36 -0700 Subject: [VoIP] VT1005 config tools In-Reply-To: <20070530143228.A1A18523.B9EE29B4@172.16.1.38> References: <20070530143228.A1A18523.B9EE29B4@172.16.1.38> Message-ID: <465DD064.5060300@josephson.com> I have answered our new friend in Argentina. Any regular participant here who has the address for these files on my server, feel free to give to others who need it under the same conditions - don't post it on the net. Cheers David J From greg at vyger.net Wed May 30 19:58:15 2007 From: greg at vyger.net (Greg Blakely) Date: Wed, 30 May 2007 19:58:15 -0500 Subject: [VoIP] Numbering - existing use of 311 Message-ID: > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of John Novack > > Here in Berkeley county WV, Verizon has 7 digit local > INTERstate dialing to MD 301, BUT, if you need to call the > overlay 240, you need to dial 11 digits. > The MD part of the LATA requires 10 digit for ALL local > calls, and 1 plus for toll calls. > Yuck! I'm glad I live here in stodgy old Minnesota! Here, in the Minneapolis / St. Paul metro calling area, we have local dialing into 6 NPAs, and all but one of those NPAs also have NXXes that are NOT part of the metro dialing area. But the dialing is pretty simple. Inside your own area code, you can dial either 7-digits or 10-digits. Local calls into any of the other five area codes is just a 10-digit call. And, to top it off, toll calls are all 1+NPA+NXXXXXX. Gee, that doesn't really sound that simple. But it is, if you're used to it. - Greg - PS. To head off the inevitable question, the 6 NPAs are 320, 507, 612, 651, 763, and 952. All but 507 and 320 **used** to be part of 612 when I moved here 13 years ago. From chad at maine.edu Wed May 30 21:57:49 2007 From: chad at maine.edu (Chad Perkins) Date: Wed, 30 May 2007 22:57:49 -0400 Subject: [VoIP] Valid CLID Message-ID: <465E016D.31748.B5CB6E2@localhost> > Opps first part was the brain fart. > > The second part sounds good to me but I think maybe there > should be more input from some of the others. > > Dennis Hock I've listened to the chatter on this over last couple weeks (via the web) while I got my e-mail and computer fixed. I tend to agree with John N's basic line of thinking; good CNET etiquette says tandems (and other clients) should always transmit *meaningful* CallerID. One should note that may not translate into a CNET dialable number, i.e. a call coming in through a portal might appear as 603-806-1499. Meaningful yes, dialable no (most CC 1 folks would have to dial it from their