From hockd at dteenergy.com Fri Jun 1 04:01:44 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 05:01:44 -0400 Subject: [VoIP] Cisco 3810 Message-ID: Lee and others. the person I purchased the 3810 from was Chris at West tech recyclers out west. He is at chrisk at westechrecyclers.com. A short summary of the unit shows it has 1 serial interface, I channelized T1/PRI port, 256K non volatile config memory, 16384K bytes of processor board Sys flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 serial (sync/async) network interface. It also has a combo Analog Voice Module containign 4 FXS and 2 FXO. These folks were very easy to do business with. I purchased two units ( one for me, one for a fellow at work). The only problem we had so far was they seemed to have lost the order in their shipping dept. It was however very quickly made right when I hadn't received the units and inquired. It sounded like they had this package and had lost the address it was to be sent to. I would have no problem using them again. At the time he had a selection of these and of course other units to choose from. Hope this helps all. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches , jnovack at stromberg-carlson.org From: john jones Sent by: voip-bounces at ckts.info Date: 05/31/2007 07:39PM Subject: Re: [VoIP] Cisco 3810 The FXO port supports DTMF, MF, and pulse dialing. On one of my 3810's, ( I have routers like you guys have phones) has a FXO port in slot 1/6. To change the out-pulsing mode, use the following command BatPhone(config-voiceport)#dial BatPhone(config-voiceport)#dial-type ? dtmf touch-tone dialer mf mf-tone dialer pulse pulse dialer Just pick the one you want. The E&M interface is exactly the same. The FXS port supports DTMF and pulse dialing. I've never tried MF on an FXS port and I'm not going to try it now! John ----- Original Message ---- From: Lee Spenadel To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Sent: Thursday, May 31, 2007 7:14:56 AM Subject: Re: [VoIP] Cisco 3810 So was I. How much was it? I'm wondering if that would be a nice device to use for FXO in lieu of the Adtran given the outpulsing problem toward my SxS. L -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Thursday, May 31, 2007 12:10 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 I see Russ Price is awake this late to enjoy my Cisco! JN John Novack wrote: > On a lighter note, with John Jones expert help, I have been working a > little with the Cisco 3810 > Reloaded with SIP software, the box can be equipped with 6 ports, in any > mix, of FXS, FXO or E&M circuits. > Paul Wills is using one for his switch and ANI. > I have mine on line right now set up for several foreign Ringback tones. > Though it may not be available at all times, give it a call at 666-3101 > through 3104 for Cisco's electronic version of some other than North > American precise tones. UK, RU, EG and JP > No phones connected, so one can call at odd hours. > > 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/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 04:03:16 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 05:03:16 -0400 Subject: [VoIP] Cisco 3810 Message-ID: In case anyone needs it we were able to pull down the docs from the Cisco site as the 3810 is a discontinued product. I can send it as an attachment I think to anyone who has one and needs the manual but can't find it on the Cisco site. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches , jnovack at stromberg-carlson.org From: john jones Sent by: voip-bounces at ckts.info Date: 05/31/2007 07:39PM Subject: Re: [VoIP] Cisco 3810 The FXO port supports DTMF, MF, and pulse dialing. On one of my 3810's, ( I have routers like you guys have phones) has a FXO port in slot 1/6. To change the out-pulsing mode, use the following command BatPhone(config-voiceport)#dial BatPhone(config-voiceport)#dial-type ? dtmf touch-tone dialer mf mf-tone dialer pulse pulse dialer Just pick the one you want. The E&M interface is exactly the same. The FXS port supports DTMF and pulse dialing. I've never tried MF on an FXS port and I'm not going to try it now! John ----- Original Message ---- From: Lee Spenadel To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Sent: Thursday, May 31, 2007 7:14:56 AM Subject: Re: [VoIP] Cisco 3810 So was I. How much was it? I'm wondering if that would be a nice device to use for FXO in lieu of the Adtran given the outpulsing problem toward my SxS. L -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Thursday, May 31, 2007 12:10 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 I see Russ Price is awake this late to enjoy my Cisco! JN John Novack wrote: > On a lighter note, with John Jones expert help, I have been working a > little with the Cisco 3810 > Reloaded with SIP software, the box can be equipped with 6 ports, in any > mix, of FXS, FXO or E&M circuits. > Paul Wills is using one for his switch and ANI. > I have mine on line right now set up for several foreign Ringback tones. > Though it may not be available at all times, give it a call at 666-3101 > through 3104 for Cisco's electronic version of some other than North > American precise tones. UK, RU, EG and JP > No phones connected, so one can call at odd hours. > > 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/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 04:33:03 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 05:33:03 -0400 Subject: [VoIP] In-Band Signaling Message-ID: Greg sounds interesting. Like you I am having way too much fun at work with of all things work (not). Too much work and not enough time. My commute has gone to 5 hours a day due to major construction on I94. Two hours in the morning and three hours at night. Eat go to bed and get up to do it again. Hopefully I'll have some time to continue to play a little soon. Take care, Dennis hock -----voip-bounces at ckts.info wrote: ----- To: "Voice Over IP Tandem for Analog Switches" From: "Greg Blakely" Sent by: voip-bounces at ckts.info Date: 05/31/2007 07:45PM Subject: [VoIP] In-Band Signaling I stumbled across an odd thing today while going through the trunk groups that attach to the Siemens EWSD switch that I work on in my real life -- a trunk group that goes to the MPLSMNDT12T toll tandem, and it's set up for MF. I was pretty busy, but I'm almost certain that we don't actually have a **live** MF trunk group going to Qwest, but it'd sure be cool if we did. The portal number I have listed on ckts.info is served by the EWSD, so, if I can get the MF trunks turned up without the need of a service order, we may have some 'production' inband signaling for all of us to listen to. Probably not, though. But, you never know _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 04:39:02 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 05:39:02 -0400 Subject: [VoIP] Cisco 3810 Message-ID: Johnyou sure are correct I posted the specs on the unit I bought. I had spoke to John some time ago but havcen't gotten back to it of course. Hey can you tell me if you can reach me at my office MW? 1 269-4996 or 0002. I think I reuped on the Dynamic DNS site but am not sure it really took. Hope you're doing well and everyone is doing good. Dennis -----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: 05/31/2007 10:15AM Subject: Re: [VoIP] Cisco 3810 I was guided by John Jones, since there are MANY flavors of the 3810 equipped with different sizes of memory, flash and option cards. If you don't have the daughter board that holds up to 6 FXO,FXS or E&M modules, 16k of flash and 64K of main memory ( I THINK I have all of that correct ) then you can't do what is needed. Those USUALLY go for 75-100 bucks with six ports equipped . There are a few on eBay now but higher price. JJ provided me with the SIP OS to replace the one born in the machine, and much hand holding to get to where I am so far. I have yet to configure the FXO circuits and test them. The 3810 normally comes equipped with an AC power supply. There was a 48 VDC power supply, but not as many of them show up Since it is SIP, the patches provided by Max and Russ don't apply. Paul Wills is using one of these along with a Linksys wireless router as his complete CNET interface. Dennis D Hock wrote: > Lee, > > I purchased one a short time ago, but havent done anything with it yet. I > think it was right around $110 or 20 can't recall for sure. It looks to be > in very nice shape, it wasfrom that fellow that was shown on here from > Ebay. > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: , "'Voice Over IP Tandem for Analog > Switches'" > From: "Lee Spenadel" > Sent by: voip-bounces at ckts.info > Date: 05/31/2007 07:14AM > Subject: Re: [VoIP] Cisco 3810 > > So was I. How much was it? I'm wondering if that would be a nice device > to > use for FXO in lieu of the Adtran given the outpulsing problem toward my > SxS. > > L > > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of > John Novack > Sent: Thursday, May 31, 2007 12:10 AM > To: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Cisco 3810 > > I see Russ Price is awake this late to enjoy my Cisco! > > JN > > > John Novack wrote: > >> On a lighter note, with John Jones expert help, I have been working a >> little with the Cisco 3810 >> Reloaded with SIP software, the box can be equipped with 6 ports, in any >> mix, of FXS, FXO or E&M circuits. >> Paul Wills is using one for his switch and ANI. >> I have mine on line right now set up for several foreign Ringback tones. >> Though it may not be available at all times, give it a call at 666-3101 >> through 3104 for Cisco's electronic version of some other than North >> American precise tones. UK, RU, EG and JP >> No phones connected, so one can call at odd hours. >> >> 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/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/ > > _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 04:59:41 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 05:59:41 -0400 Subject: [VoIP] Numbering - existing use of 311 Message-ID: That could very well be. Depends on the local calling area and where one is at. Also seems like Verizon (GTE) was allowed to implement slightly different than att/SBC/Ameritech. All in all a bit of a headache. Seems like it would be much easier to just move to ten digit dialing and be done with it. Or a the very least allow it to complete if it should and not block it based on some politcal decision. The reason fro not iallowing the 1 and ten to complete according tot eh PSC is that everyone knows when you dial 1 it is long distance and this sends the wrong information to the customer. So better to block it. 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: 05/31/2007 09:37AM Subject: Re: [VoIP] Numbering - existing use of 311 I could be mistaken but I think I my sister said that in the 989 NPA where she lives there are both 10 and 11 digit calls within the immediate surrounding part of the Saginaw-Bay City-Midland area. Sounds like what you are describing. She might have even said that there were cases where there were both 10 and 11 digit calls to the same exchange. I'm not sure about this second part. If that seems confusing, when they split eastern Long Island (Suffolk County) from 516 to 631 some years ago, they divided the 420 exchange serving Farmingdale into a 516 and 631 part. Farmingdale straddles the Nassau-Suffolk County line. Calls from outside the area only complete with the correct area code. So it was impossible to determine which area code to dial. One person I knew stayed with 516-420 while another ended up with a 631-420 number. People within the same exchange have to dial the other area code to reach people on the other side of the county line. Maybe the county line runs through the CO building and they assigned all line equipments on one side of the line to 516 and those on the other to 631, reassigning people to different line equipment to match the geographical location of their line equipment with the side of the county line they are on. Nothing would surprise me. SK Dennis D Hock wrote: > I am not sure but suspect as John and others have indicated I'll bet it is > working somewhere in the vast NANP. In Michigan where I live we have > received extended local calling to any adjacent exchange that touches > yours. Now one would think that would mean you could dial it as a seven > digit number. Nope guess again, and you can't dial it as a 1 plus ten > digit. It can only be dialed as ten digits. You think I would remember > when calling the Home Depot in the next town over but almost every time it > takes me two or three tries to get it right. > > Dennis Hock > > -----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: 05/30/2007 08:18AM > Subject: Re: [VoIP] Numbering - existing use of 311 > > 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/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/ > > _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 05:14:35 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 06:14:35 -0400 Subject: [VoIP] Valid CLID Message-ID: I don't see how either of you could resemble that remark as you are both from the industry. The politicians have nothing better to do and certainly have shown over time they don't understand anything more technical than changing the batteries in a flashlight. Beyond that they only see dollars or the hope of them. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: , "Voice Over IP Tandem for Analog Switches" From: "Greg Blakely" Sent by: voip-bounces at ckts.info Date: 05/31/2007 10:03AM Subject: Re: [VoIP] Valid CLID > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] > On Behalf Of John Novack > Sent: Wednesday, May 30, 2007 10:54 PM > To: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Valid CLID > > used to be normal NANP, before all the political types stuck > their fingers in the mix. > Harumph! I resemble that remark... _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 05:26:29 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 06:26:29 -0400 Subject: [VoIP] Valid CLID Message-ID: I agree with John. I understand or at least I think I understand why greg and some others may wish or need to use "200" as a NPA, and I do recall gregs post but frankly and honestly I haven't done a thing to my dial plan to accomodate that. Not that I am against it but I am against it at this time. I would say Johns comment of KISS should apply. I don't think we want to do some things like we find ourselves in here at work. That is "ICTIP" or In Confusion There Is Profit ;-). I think its day may come that we will need a NPA but I don't think it is necessary to have it yet. Just my opinion. 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: 05/30/2007 11:53PM Subject: Re: [VoIP] Valid CLID Chad Perkins wrote: > > To use '200' or not to use '200'. The 200 area code has been in CNET long before I was (as part of the dial plan samples). Some of us need it to route outgoing calls that otherwise would be routed to the PSTN by Asterisk, I route calls out of CNET by dialing 9-1-XXX-XXXX when I use my Stanaphone account, and 8-1-XXX-XXXX when I use Gizmo. Since I access Asterisk via several phones, either my house system Panasonic 1232, my EM switch via trunk circuits, and SIP phones, my situation may differ from others, but I sure as hell don't want to dial ALL CNET calls with 10 digits. > and most of us I think strip it off just before sending the called number to ENUM. I have made NO provisions for using nor receiving the 200 NPA. I dial CNET calls as 7 digits for NA, and Asterisk adds the 1 towards the network. FOr other than NA calls, I dial 011. To my mind, this is about as close as I can get to the what used to be normal NANP, before all the political types stuck their fingers in the mix. Asterisk can certainly be programmed around this, but it seems silly to me right now to have an NPA long before we need it, if ever. Several of us have multiple office codes that could be released, or have been released for others to use. Office codes can easily be divided up between two distant switches, if necessary. Lets Keep It Simple - KISS John Novack _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From lee at spenadel.com Fri Jun 1 06:49:22 2007 From: lee at spenadel.com (Lee Spenadel) Date: Fri, 1 Jun 2007 07:49:22 -0400 Subject: [VoIP] Adtran outpulsing In-Reply-To: <20070601030501.71604.qmail@web51603.mail.re2.yahoo.com> References: <000a01c7a37e$24e23600$6ea6a200$@com> <20070601030501.71604.qmail@web51603.mail.re2.yahoo.com> Message-ID: <00ac01c7a442$e5c04b90$b140e2b0$@com> Max, Excellent thought(s) and quite logical. I use a Digium T400P clone of some sort - I'd have to look to see who made it. I can't remember what Russ is using, but maybe that can shed some light if he's using a different board. I know he was using an Adtran like me with the same FXO. Lee -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of ikjtel Sent: Thursday, May 31, 2007 11:05 PM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Adtran outpulsing FWIW, the X100P uses a different low-level driver (wcfxo.c) than channel-bank ports use (wct1xxp.c et al.), and the Digium TDM400 uses yet another one (wctdm.c) although all of these in turn rely upon zaptel.c for the generic processing... Then again SIP has yet another driver (and is even more different in that user-space [asterisk] speaks to SIP devices differently - chan_sip.c instead of chan_zap.c) All clear??? :-) It is easy to hear the irregularity in the outpulsing (at least it was some time ago when dialling thru Lee's system which had the "hearpulsing" patch applied)... The way in which the signalling is conveyed in the Adtran-FXO case is by way of the "RBS bits" (robbed-bit signalling) using the A and B bits in the T1 frames. Inasmuch as the X100P's seem to work OK, it is tempting to rule out zaptel.c as being the culprit. This leaves wct[e]xxxp.c, the T1 itself and the card, and the Adtran as possible suspects. One would *hope* that the Adtran would be somewhat more robust than to fall into such a stupid bug, but it's possible I suppose. The T1 and card are too stupid to be prime suspects, but again it's possible. By amassing more data points we might be able to remove one or more of the remaining suspects from the list. I think it's probably safe to say that SIP is a different enough beast that it's not in play in this elimination excercise. However, if someone has a genuine TDM400 with FXO's (probably the Varion "clones" would only add to the confusion here) it might provide an interesting data point as to whether they have the irregularity or not... Likewise, if someone has another brand of channel bank FXO other than an Adtran one it would be *very* interesting to see if the problem occurs there as well... (My own unsubstantiated suspicion is that it would). Unfortunately my channel banks only have FXS, and the one time I looked to see how hard it might be to rig up a test environment, it seemed to be unfeasible. It didn't appear that the signalling bits worked the same way (or similarly enough) ... More data / feedback is welcome, of course... Best Max p.s. One *could* open a trouble ticket with Asterisk, I suppose ;-) --- Lee Spenadel wrote: > The Adtran problem may not be apparent in all > situations. Russ Price did > some testing after my reported problems with > consistent dial pulsing from > the Adtran FXO to my SxS. Russ' and my switches > were somewhat intolerant of > the timing of the Adtran pulsing. Russ tried > modifying the pulsing > make/break times in asterisk with no change in > pulsing to his SxS. In my > case it lead to mis-dialing of extensions. It > doesn't do this all of the > time. It's somewhat intermittent. When I placed a > WildCard X100P in my > Asterisk box and set it up to pulse to my SxS, all > worked fine. > > I guess the bottom line is that the Adtran FXO does > pulse out the digits, > but depending on the tolerances of your switch, you > may or may not have a > problem. > > Lee > > -----Original Message----- > From: voip-bounces at ckts.info > [mailto:voip-bounces at ckts.info] On Behalf Of > Jonathan Kay > Sent: Thursday, May 31, 2007 7:41 AM > To: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Adtran (was Cisco 3810) > > Lee Spenadel wrote: > > So was I. How much was it? I'm wondering if that > would be a nice device > to > > use for FXO in lieu of the Adtran given the > outpulsing problem toward my > > SxS. > > > > > > Is there a definite problem with Adtran channel > banks and rotary dialling? > I'm more than a bit hacked off if that's the case, > after paying loads of > shipping to the UK and big bucks to obtain an FXO > loaded Adtran. > I've still not had time to set the thing up so far. > > Jon K > _______________________________________________ > 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/ > ____________________________________________________________________________ ________ Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 _______________________________________________ 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 Fri Jun 1 07:37:33 2007 From: jjones3601 at yahoo.com (john jones) Date: Fri, 1 Jun 2007 05:37:33 -0700 (PDT) Subject: [VoIP] Cisco 3810 Message-ID: <789999.48466.qm@web34305.mail.mud.yahoo.com> You also want to make sure you get 64M of DRAM, or factor in the cost of an upgrade if it isn't equipped that way from the beginning. John ----- Original Message ---- From: Dennis D Hock To: Voice Over IP Tandem for Analog Switches Cc: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 5:01:44 AM Subject: Re: [VoIP] Cisco 3810 Lee and others. the person I purchased the 3810 from was Chris at West tech recyclers out west. He is at chrisk at westechrecyclers.com. A short summary of the unit shows it has 1 serial interface, I channelized T1/PRI port, 256K non volatile config memory, 16384K bytes of processor board Sys flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 serial (sync/async) network interface. It also has a combo Analog Voice Module containign 4 FXS and 2 FXO. These folks were very easy to do business with. I purchased two units ( one for me, one for a fellow at work). The only problem we had so far was they seemed to have lost the order in their shipping dept. It was however very quickly made right when I hadn't received the units and inquired. It sounded like they had this package and had lost the address it was to be sent to. I would have no problem using them again. At the time he had a selection of these and of course other units to choose from. Hope this helps all. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches , jnovack at stromberg-carlson.org From: john jones Sent by: voip-bounces at ckts.info Date: 05/31/2007 07:39PM Subject: Re: [VoIP] Cisco 3810 The FXO port supports DTMF, MF, and pulse dialing. On one of my 3810's, ( I have routers like you guys have phones) has a FXO port in slot 1/6. To change the out-pulsing mode, use the following command BatPhone(config-voiceport)#dial BatPhone(config-voiceport)#dial-type ? dtmf touch-tone dialer mf mf-tone dialer pulse pulse dialer Just pick the one you want. The E&M interface is exactly the same. The FXS port supports DTMF and pulse dialing. I've never tried MF on an FXS port and I'm not going to try it now! John ----- Original Message ---- From: Lee Spenadel To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Sent: Thursday, May 31, 2007 7:14:56 AM Subject: Re: [VoIP] Cisco 3810 So was I. How much was it? I'm wondering if that would be a nice device to use for FXO in lieu of the Adtran given the outpulsing problem toward my SxS. L -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Thursday, May 31, 2007 12:10 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 I see Russ Price is awake this late to enjoy my Cisco! JN John Novack wrote: > On a lighter note, with John Jones expert help, I have been working a > little with the Cisco 3810 > Reloaded with SIP software, the box can be equipped with 6 ports, in any > mix, of FXS, FXO or E&M circuits. > Paul Wills is using one for his switch and ANI. > I have mine on line right now set up for several foreign Ringback tones. > Though it may not be available at all times, give it a call at 666-3101 > through 3104 for Cisco's electronic version of some other than North > American precise tones. UK, RU, EG and JP > No phones connected, so one can call at odd hours. > > 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/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From jjones3601 at yahoo.com Fri Jun 1 07:37:33 2007 From: jjones3601 at yahoo.com (john jones) Date: Fri, 1 Jun 2007 05:37:33 -0700 (PDT) Subject: [VoIP] Cisco 3810 Message-ID: <789999.48466.qm@web34305.mail.mud.yahoo.com> You also want to make sure you get 64M of DRAM, or factor in the cost of an upgrade if it isn't equipped that way from the beginning. John ----- Original Message ---- From: Dennis D Hock To: Voice Over IP Tandem for Analog Switches Cc: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 5:01:44 AM Subject: Re: [VoIP] Cisco 3810 Lee and others. the person I purchased the 3810 from was Chris at West tech recyclers out west. He is at chrisk at westechrecyclers.com. A short summary of the unit shows it has 1 serial interface, I channelized T1/PRI port, 256K non volatile config memory, 16384K bytes of processor board Sys flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 serial (sync/async) network interface. It also has a combo Analog Voice Module containign 4 FXS and 2 FXO. These folks were very easy to do business with. I purchased two units ( one for me, one for a fellow at work). The only problem we had so far was they seemed to have lost the order in their shipping dept. It was however very quickly made right when I hadn't received the units and inquired. It sounded like they had this package and had lost the address it was to be sent to. I would have no problem using them again. At the time he had a selection of these and of course other units to choose from. Hope this helps all. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches , jnovack at stromberg-carlson.org From: john jones Sent by: voip-bounces at ckts.info Date: 05/31/2007 07:39PM Subject: Re: [VoIP] Cisco 3810 The FXO port supports DTMF, MF, and pulse dialing. On one of my 3810's, ( I have routers like you guys have phones) has a FXO port in slot 1/6. To change the out-pulsing mode, use the following command BatPhone(config-voiceport)#dial BatPhone(config-voiceport)#dial-type ? dtmf touch-tone dialer mf mf-tone dialer pulse pulse dialer Just pick the one you want. The E&M interface is exactly the same. The FXS port supports DTMF and pulse dialing. I've never tried MF on an FXS port and I'm not going to try it now! John ----- Original Message ---- From: Lee Spenadel To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Sent: Thursday, May 31, 2007 7:14:56 AM Subject: Re: [VoIP] Cisco 3810 So was I. How much was it? I'm wondering if that would be a nice device to use for FXO in lieu of the Adtran given the outpulsing problem toward my SxS. L -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Thursday, May 31, 2007 12:10 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 I see Russ Price is awake this late to enjoy my Cisco! JN John Novack wrote: > On a lighter note, with John Jones expert help, I have been working a > little with the Cisco 3810 > Reloaded with SIP software, the box can be equipped with 6 ports, in any > mix, of FXS, FXO or E&M circuits. > Paul Wills is using one for his switch and ANI. > I have mine on line right now set up for several foreign Ringback tones. > Though it may not be available at all times, give it a call at 666-3101 > through 3104 for Cisco's electronic version of some other than North > American precise tones. UK, RU, EG and JP > No phones connected, so one can call at odd hours. > > 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/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From jnovack at stromberg-carlson.org Fri Jun 1 08:21:21 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Fri, 01 Jun 2007 09:21:21 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: References: Message-ID: <46601D51.60507@stromberg-carlson.org> Dennis D Hock wrote: > The reason for not iallowing the 1 and ten to complete according to the PSC is that everyone knows when you dial 1 it is long distance and this sends the wrong information to the > customer. So better to block it. > > Dennis H. > Except in IL, where 1 plus is REQUIRED for ALL 10 digit ( 11 digit ) calls Probably other states as well. What we get when technical decisions are allowed to be made by bureaucrats. John Novack From hockd at dteenergy.com Fri Jun 1 08:48:46 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 09:48:46 -0400 Subject: [VoIP] Cisco 3810 Message-ID: John, that is to say I don't have it??? Any idea the cost?? ballpark Thanks, Dennis H -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: john jones Sent by: voip-bounces at ckts.info Date: 06/01/2007 08:37AM cc: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 You also want to make sure you get 64M of DRAM, or factor in the cost of an upgrade if it isn't equipped that way from the beginning. John ----- Original Message ---- From: Dennis D Hock To: Voice Over IP Tandem for Analog Switches Cc: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 5:01:44 AM Subject: Re: [VoIP] Cisco 3810 Lee and others. the person I purchased the 3810 from was Chris at West tech recyclers out west. He is at chrisk at westechrecyclers.com. A short summary of the unit shows it has 1 serial interface, I channelized T1/PRI port, 256K non volatile config memory, 16384K bytes of processor board Sys flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 serial (sync/async) network interface. It also has a combo Analog Voice Module containign 4 FXS and 2 FXO. These folks were very easy to do business with. I purchased two units ( one for me, one for a fellow at work). The only problem we had so far was they seemed to have lost the order in their shipping dept. It was however very quickly made right when I hadn't received the units and inquired. It sounded like they had this package and had lost the address it was to be sent to. I would have no problem using them again. At the time he had a selection of these and of course other units to choose from. Hope this helps all. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches , jnovack at stromberg-carlson.org From: john jones Sent by: voip-bounces at ckts.info Date: 05/31/2007 07:39PM Subject: Re: [VoIP] Cisco 3810 The FXO port supports DTMF, MF, and pulse dialing. On one of my 3810's, ( I have routers like you guys have phones) has a FXO port in slot 1/6. To change the out-pulsing mode, use the following command BatPhone(config-voiceport)#dial BatPhone(config-voiceport)#dial-type ? dtmf touch-tone dialer mf mf-tone dialer pulse pulse dialer Just pick the one you want. The E&M interface is exactly the same. The FXS port supports DTMF and pulse dialing. I've never tried MF on an FXS port and I'm not going to try it now! John ----- Original Message ---- From: Lee Spenadel To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Sent: Thursday, May 31, 2007 7:14:56 AM Subject: Re: [VoIP] Cisco 3810 So was I. How much was it? I'm wondering if that would be a nice device to use for FXO in lieu of the Adtran given the outpulsing problem toward my SxS. L -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Thursday, May 31, 2007 12:10 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 I see Russ Price is awake this late to enjoy my Cisco! JN John Novack wrote: > On a lighter note, with John Jones expert help, I have been working a > little with the Cisco 3810 > Reloaded with SIP software, the box can be equipped with 6 ports, in any > mix, of FXS, FXO or E&M circuits. > Paul Wills is using one for his switch and ANI. > I have mine on line right now set up for several foreign Ringback tones. > Though it may not be available at all times, give it a call at 666-3101 > through 3104 for Cisco's electronic version of some other than North > American precise tones. UK, RU, EG and JP > No phones connected, so one can call at odd hours. > > 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/ _______________________________________________ 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/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 08:50:53 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 09:50:53 -0400 Subject: [VoIP] Numbering - existing use of 311 Message-ID: Absolutely John. I should have qulified that to say that was the explanation from our Michigan PSC. Others as you said are no doubt looking at it from their rose colored glasses and implementing accordingly. Dennis -----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: 06/01/2007 09:21AM Subject: Re: [VoIP] Numbering - existing use of 311 Dennis D Hock wrote: > The reason for not iallowing the 1 and ten to complete according to the PSC is that everyone knows when you dial 1 it is long distance and this sends the wrong information to the > customer. So better to block it. > > Dennis H. > Except in IL, where 1 plus is REQUIRED for ALL 10 digit ( 11 digit ) calls Probably other states as well. What we get when technical decisions are allowed to be made by bureaucrats. John Novack _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From bbj at innismir.net Fri Jun 1 08:53:47 2007 From: bbj at innismir.net (Ben Jackson) Date: Fri, 01 Jun 2007 09:53:47 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <46601D51.60507@stromberg-carlson.org> References: <46601D51.60507@stromberg-carlson.org> Message-ID: <466024EB.3020403@innismir.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 John Novack wrote: > > Dennis D Hock wrote: >> The reason for not iallowing the 1 and ten to complete according to >> the PSC is that everyone knows when you dial 1 it is long distance >> and this sends the wrong information to the customer. So better to >> block it. >> >> Dennis H. >> > Except in IL, where 1 plus is REQUIRED for ALL 10 digit ( 11 digit ) > calls Probably other states as well. What we get when technical > decisions are allowed to be made by bureaucrats. > Similar in MA. When the overlay codes came into being, Verizon "preferred" you dial +1 for all calls local or toll. - -- Ben Jackson - N1WBV - New Bedford, MA bbj innismir.net - http://www.innismir.net/ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRmAk6wQiWVsfSvVvAQLhzwgAmwV7j6IzKS+meeNZK421IWvY7Vq7gLdL rlyWLV9HMCez3HEetArBOzXa/kuQmfNQI9DoZIWkHQw4dZYULCZQ+CJuvv2OBbjF zl0He10bkSwO9L0dZwTbZn4pveyT3Mv3uKNG8nQSnWMYpU4chJKWO+N6pIspsSxd QuJ+ZyG7/XJPhJffKZLAsRhLjNoFvAE3UBsxYUCMXvXq1dcGS1tt8o0GF4VHiNyW DXojp9zBeAEY/HMQOSMoP+j9RqOY9jKYesY2aZLr5ZtoN6vlDUAqsCzHjRzt71EC X5uBTJsMHSy56OWvXi6jcZcFWSovV4Y2FrO1hLBQbUD3y7KLo9hRoA== =ZyRw -----END PGP SIGNATURE----- From rdekema at gmail.com Fri Jun 1 09:26:09 2007 From: rdekema at gmail.com (Rusty Dekema) Date: Fri, 1 Jun 2007 10:26:09 -0400 Subject: [VoIP] Cisco 3810 In-Reply-To: References: Message-ID: <68171c120706010726o54c534aar5af2965e6ddef461@mail.gmail.com> On 6/1/07, Dennis D Hock wrote: > John, that is to say I don't have it??? Any idea the cost?? ballpark > > Thanks, > > Dennis H Looks like about $30 on eBay. I forget exactly what kind of memory is needed for the MC3810, I think it's a parity SIMM. However, if you search for mc3810 on eBay, there are a ton of listings of "64M Memory Module for MC3810" for about $30 each. I'm pretty sure you need the 64M of RAM to run any version of IOS that supports SIP on the 3810, but I could be wrong about that. Rusty From jnovack at stromberg-carlson.org Fri Jun 1 10:14:57 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Fri, 01 Jun 2007 11:14:57 -0400 Subject: [VoIP] Cisco 3810 In-Reply-To: <68171c120706010726o54c534aar5af2965e6ddef461@mail.gmail.com> References: <68171c120706010726o54c534aar5af2965e6ddef461@mail.gmail.com> Message-ID: <466037F1.6070402@stromberg-carlson.org> I am also a LITTLE confused about all of this The unit I ( sometimes ) have working has 2 replaceable memory modules One is marked SMART and then a long part number Is this the flash memory? The NVram? Another, shorter module is marked 64MB sync 125Mhz This is the main memory? Clearly one needs the 16 meg flash as the SIP image is 13 meg All old files have to be offloaded and deleted before the SIP image can be loaded There are often 3810's that show up on eBay, but not all will allow this configuration Not all have all the cards needed either, and purchasing the AVM daughterboard alone is $60, then up to 6 modules. Finding one properly equipped can often be the lowest cost. John Novack Rusty Dekema wrote: > On 6/1/07, Dennis D Hock wrote: > >> John, that is to say I don't have it??? Any idea the cost?? ballpark >> >> Thanks, >> >> Dennis H >> > > Looks like about $30 on eBay. I forget exactly what kind of memory is > needed for the MC3810, I think it's a parity SIMM. However, if you > search for mc3810 on eBay, there are a ton of listings of "64M Memory > Module for MC3810" for about $30 each. > > I'm pretty sure you need the 64M of RAM to run any version of IOS that > supports SIP on the 3810, but I could be wrong about that. > > > Rusty From mark at rudholm.com Fri Jun 1 10:44:22 2007 From: mark at rudholm.com (Mark Rudholm) Date: Fri, 01 Jun 2007 08:44:22 -0700 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: References: Message-ID: <46603ED6.9050500@rudholm.com> Maybe I have a southern California bias or something, but I never really understood the whole "1+ Toll Alerting" business. I realize some people love it, but I always thought it was silly, and now that long-distance is down in the 0-4 cents per minute range, is it really that big of a deal? Correct me if I'm wrong, but wasn't 1+ dialing introduced to disambiguate out-of-npa calls when generalized (from NNX to NXX) office codes were first assigned back in the early 70s? Back when NPAs were all N0/1X and office codes were all NNX, there were no O.C./npa collisions, so all calls could be dialed as 7D or 10D without ambiguity. So the idea of 1+ = toll alerting seems like an artifact, not something by design. Living in Los Angeles, where area codes are quite numerous, people have generally started just dialing 1+10D for all calls. Given that, it seems to me that the 1 is now just anachronistic baggage, so for my systems, I've dropped it. I just use 10D for all calls. You can use 7D but there's a timeout. The telcos still require the 1, however. Also, in 310, which is scheduled for an overlay, 1+10D is now mandatory on *all* calls. I understand the rationale here is that since the CLECs will be the ones primarily getting assignments in the new area code, the CPUC wanted to level they playing field by requiring 11D, thereby eliminating the preferability of 310 office codes. But anyway, meh on 1+ toll alerting, I say. Dennis D Hock wrote: > Absolutely John. I should have qulified that to say that was the > explanation from our Michigan PSC. Others as you said are no doubt looking > at it from their rose colored glasses and implementing accordingly. > > Dennis > > -----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: 06/01/2007 09:21AM > Subject: Re: [VoIP] Numbering - existing use of 311 > > > > Dennis D Hock wrote: > >> The reason for not iallowing the 1 and ten to complete according to the > PSC is that everyone knows when you dial 1 it is long distance and this > sends the wrong information to the >> customer. So better to block it. >> >> Dennis H. >> > Except in IL, where 1 plus is REQUIRED for ALL 10 digit ( 11 digit ) calls > Probably other states as well. > What we get when technical decisions are allowed to be made by bureaucrats. > > 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/ From stfkerman at jps.net Fri Jun 1 11:01:44 2007 From: stfkerman at jps.net (Steph Kerman) Date: Fri, 01 Jun 2007 12:01:44 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <46603ED6.9050500@rudholm.com> References: <46603ED6.9050500@rudholm.com> Message-ID: <466042E8.2010601@jps.net> Areas that used 10D dialing before NXX office codes were introduced were served by Panel, Crossbar or ESS. These did indeed adopt the use of 1+ to enable the CO to know whether to wait for 7 or 10 digits. However 1+10D dialing was always required in most areas served by SXS equipment to immediately pass the call through to the toll office when a 10 digit call was dialed. So it is not correct that when all NPAs were N0/1X all calls were dialed as 7D or 10D. Many SXS areas, in addition to requiring 1+10D dilaing also required 1+7D dialing for short haul toll, for the same reason: to allow the toll office to capture the dialed digits for toll billing. Whether the 1 prefix was a requirement for regulatory reasons, it was a necessity for equipment reasons. 1+ dialing was also sometimes used on common control offices in areas with mixed SXS and common control equipment to achieve uniform dialing practices. But many areas that were served by common control equipment used 7 digit dialing for local as well as short haul toll, without a 1+ prefix to distinguish toll and local. Steph Mark Rudholm wrote: > Maybe I have a southern California bias or something, but I never > really understood the whole "1+ Toll Alerting" business. I realize > some people love it, but I always thought it was silly, and now > that long-distance is down in the 0-4 cents per minute range, is > it really that big of a deal? > > Correct me if I'm wrong, but wasn't 1+ dialing introduced to > disambiguate out-of-npa calls when generalized (from NNX to NXX) > office codes were first assigned back in the early 70s? Back when NPAs > were all N0/1X and office codes were all NNX, there were no O.C./npa > collisions, so all calls could be dialed as 7D or 10D without > ambiguity. So the idea of 1+ = toll alerting seems like an artifact, > not something by design. > > Living in Los Angeles, where area codes are quite numerous, people > have generally started just dialing 1+10D for all calls. Given that, > it seems to me that the 1 is now just anachronistic baggage, so for > my systems, I've dropped it. I just use 10D for all calls. You > can use 7D but there's a timeout. > > The telcos still require the 1, however. > > Also, in 310, which is scheduled for an overlay, 1+10D is now > mandatory on *all* calls. I understand the rationale here is that > since the CLECs will be the ones primarily getting assignments in > the new area code, the CPUC wanted to level they playing field by > requiring 11D, thereby eliminating the preferability of 310 office > codes. > > But anyway, meh on 1+ toll alerting, I say. > > Dennis D Hock wrote: > >> Absolutely John. I should have qulified that to say that was the >> explanation from our Michigan PSC. Others as you said are no doubt looking >> at it from their rose colored glasses and implementing accordingly. >> >> Dennis >> >> -----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: 06/01/2007 09:21AM >> Subject: Re: [VoIP] Numbering - existing use of 311 >> >> >> >> Dennis D Hock wrote: >> >> >>> The reason for not iallowing the 1 and ten to complete according to the >>> >> PSC is that everyone knows when you dial 1 it is long distance and this >> sends the wrong information to the >> >>> customer. So better to block it. >>> >>> Dennis H. >>> >>> >> Except in IL, where 1 plus is REQUIRED for ALL 10 digit ( 11 digit ) calls >> Probably other states as well. >> What we get when technical decisions are allowed to be made by bureaucrats. >> >> 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/ >> > > _______________________________________________ > 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 Fri Jun 1 11:22:56 2007 From: stfkerman at jps.net (Steph Kerman) Date: Fri, 01 Jun 2007 12:22:56 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: References: Message-ID: <466047E0.3040902@jps.net> Here in NYC we have to dial 1+ for everything, local or toll. Originally, 212 served all 5 boroughs. In the 80s they moved Brooklyn, Queens and Staten Island to 718. After the remaining crossbar was retired, they shifted the Bronx from 212 to 718, leaving 212 for Manhattan only. 917 was assigned for pagers only throughout the entire area. At this point 212 is overlayed with 646 and 347. 917 is now partially overlayed on 212 although it's mostly used for cellphones. I'm not sure whether 347 and 917 are overlayed on 718 too or only overlayed on 212. There does not seem to be a simple way to find out without a lot of study of Telcodata listings. It's a mess... Steph Dennis D Hock wrote: > That could very well be. Depends on the local calling area and where one > is at. Also seems like Verizon (GTE) was allowed to implement slightly > different than att/SBC/Ameritech. All in all a bit of a headache. Seems > like it would be much easier to just move to ten digit dialing and be done > with it. Or a the very least allow it to complete if it should and not > block it based on some politcal decision. The reason fro not iallowing the > 1 and ten to complete according tot eh PSC is that everyone knows when you > dial 1 it is long distance and this sends the wrong information to the > customer. So better to block it. > > 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: 05/31/2007 09:37AM > Subject: Re: [VoIP] Numbering - existing use of 311 > > I could be mistaken but I think I my sister said that in the 989 NPA > where she lives there are both 10 and 11 digit calls within the > immediate surrounding part of the Saginaw-Bay City-Midland area. Sounds > like what you are describing. She might have even said that there were > cases where there were both 10 and 11 digit calls to the same exchange. > I'm not sure about this second part. > > If that seems confusing, when they split eastern Long Island (Suffolk > County) from 516 to 631 some years ago, they divided the 420 exchange > serving Farmingdale into a 516 and 631 part. Farmingdale straddles the > Nassau-Suffolk County line. Calls from outside the area only complete > with the correct area code. So it was impossible to determine which > area code to dial. One person I knew stayed with 516-420 while another > ended up with a 631-420 number. People within the same exchange have > to dial the other area code to reach people on the other side of the > county line. > > Maybe the county line runs through the CO building and they assigned all > line equipments on one side of the line to 516 and those on the other to > 631, reassigning people to different line equipment to match the > geographical location of their line equipment with the side of the > county line they are on. Nothing would surprise me. > > SK > > Dennis D Hock wrote: > >> I am not sure but suspect as John and others have indicated I'll bet it >> > is > >> working somewhere in the vast NANP. In Michigan where I live we have >> received extended local calling to any adjacent exchange that touches >> yours. Now one would think that would mean you could dial it as a seven >> digit number. Nope guess again, and you can't dial it as a 1 plus ten >> digit. It can only be dialed as ten digits. You think I would remember >> when calling the Home Depot in the next town over but almost every time >> > it > >> takes me two or three tries to get it right. >> >> Dennis Hock >> >> -----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: 05/30/2007 08:18AM >> Subject: Re: [VoIP] Numbering - existing use of 311 >> >> 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/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/ >> >> >> > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voipProject Web Page: > http://www.ckts.info/ > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From jnovack at stromberg-carlson.org Fri Jun 1 11:21:37 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Fri, 01 Jun 2007 12:21:37 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <46603ED6.9050500@rudholm.com> References: <46603ED6.9050500@rudholm.com> Message-ID: <46604791.1080809@stromberg-carlson.org> Mark Rudholm wrote: > Maybe I have a southern California bias or something, but I never really understood the whole "1+ Toll Alerting" business. I realize some people love it, but I always thought it was silly, > Historically, it was pretty much a requirement in Step offices to reach the toll operator or for DDD. Some locales with party lines would dial 112x with the X designating the party on a given line In the larger Bell cities ( Except LA for other historical reasons ) there was no 1+ dialing in the EM days 211 would get one to the toll operator, then later dialing 10 digits when the NPA was in the form of N0/1X would get directed properly Things changed once we enter the electronic era and CLECs and NPA's with the form NXX So, NO, it wasn't/isn't silly > and now that long-distance is down in the 0-4 cents per minute range, is it really that big of a deal? > I suppose that all depends on your perspective , calling needs and income level! > Correct me if I'm wrong, but wasn't 1+ dialing introduced to > disambiguate out-of-npa calls when generalized (from NNX to NXX) > office codes were first assigned back in the early 70s? Back when NPAs were all N0/1X and office codes were all NNX, there were no O.C./npa collisions, so all calls could be dialed as 7D or 10D without ambiguity. Only in panel/XBAR offices. > So the idea of 1+ = toll alerting seems like an artifact, > not something by design. > See above. > Living in Los Angeles, where area codes are quite numerous, people have generally started just dialing 1+10D for all calls. LA was, in the EM era, all or almost all step, Historically due to the merging and buyout of independents, then I suspect due to the frugal nature of the PUC regarding Pacific Bell. At one point AT&T seriously wanted to rid itself of that unit due to the "frugal" nature of the California PUC. See "The Fall of the Bell System" for much more information on the corporate side of things. > Given that,it seems to me that the 1 is now just anachronistic baggage, so for my systems, I've dropped it. Yet several state PUC insist on dictating that it be used for toll, or ALL calls beyond 7 digits. > I just use 10D for all calls. You can use 7D but there's a timeout. > > The telcos still require the 1, however. > In many cases, this is dictated to them by the PUC, or whatever it is called in a given state. Part of the current insanity. In MO, for example, the PUC dictated where a split was to be located, rather than growth patterns of the population. I believe the only thing that saved SBC was the decline in lines due to HSIA and release of 2nd lines along with the decline in FAX usage. IMHO, once the US was all electronic, there should NEVER have been another split. Overlays cost everyone less, even though they require 10 digit dialing in many cases. Intelligence in the switch could be used, but it is much easier for the user to just dial 10 digits for all local calls > Also, in 310, which is scheduled for an overlay, 1+10D is now > mandatory on *all* calls. I understand the rationale here is that since the CLECs will be the ones primarily getting assignments in the new area code, the CPUC wanted to level they playing field by requiring 11D, thereby eliminating the preferability of 310 office codes. > That makes little sense. 10 digit local calls, 11 digit toll calls is really the best solution. It helps the user determine toll vs local and simplifys switch translations. Right now anyone traveling and not in the know can easily become frustrated. Add to that the cell phone dialing requirements My carrier requires 10 digits for ALL calls since my home LATA has an overlay and requires it, adds the 1 towards the network and calls go through. Since all my cell calls are billed the same, there is no toll vs local issue or time of day issue, calls to other cells in the same carrier are free, so it is simply a minute issue . The landline carriers and PUC's could learn from that. John Novack From mark at rudholm.com Fri Jun 1 11:51:23 2007 From: mark at rudholm.com (Mark Rudholm) Date: Fri, 01 Jun 2007 09:51:23 -0700 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <466042E8.2010601@jps.net> References: <46603ED6.9050500@rudholm.com> <466042E8.2010601@jps.net> Message-ID: <46604E8B.70007@rudholm.com> OK, so it was still a technical issue, not a toll alerting system for customers. So I think my opinion remains --meh on 1+ Toll Alerting. I remember going to Huntington Beach as a kid (late 70s, iirc) and thinking "hey, Lee Vining is really far away and in the same area code (back then, 714 stretched from Mexico to Lake Tahoe), I'm going to dial it as a seven digit call and see what happens" I got an intercept telling me I needed to dial a 1 first. That was General Telephone land though, so I didn't really take it seriously. Speaking of which, the other day I tried dialing some places in Arizona that are in the Los Angeles LATA (#730) from my payphone. This *should* have worked, since it was intra-LATA. Turns out some destinations work and some don't. I asked the operator why I couldn't make a coin paid call despite it being intra-LATA. She didn't really say anything useful. I suspect ACTS is just out of date. Man, I'd really like to have my own real ACTS system. I don't have the room for a DMS200 or 4ESS, though. :-) Steph Kerman wrote: > Areas that used 10D dialing before NXX office codes were introduced were > served by Panel, Crossbar or ESS. These did indeed adopt the use of 1+ > to enable the CO to know whether to wait for 7 or 10 digits. However > 1+10D dialing was always required in most areas served by SXS equipment > to immediately pass the call through to the toll office when a 10 digit > call was dialed. So it is not correct that when all NPAs were N0/1X all > calls were dialed as 7D or 10D. > > Many SXS areas, in addition to requiring 1+10D dilaing also required > 1+7D dialing for short haul toll, for the same reason: to allow the toll > office to capture the dialed digits for toll billing. Whether the 1 > prefix was a requirement for regulatory reasons, it was a necessity for > equipment reasons. 1+ dialing was also sometimes used on common control > offices in areas with mixed SXS and common control equipment to achieve > uniform dialing practices. But many areas that were served by common > control equipment used 7 digit dialing for local as well as short haul > toll, without a 1+ prefix to distinguish toll and local. > > Steph > > Mark Rudholm wrote: >> Maybe I have a southern California bias or something, but I never >> really understood the whole "1+ Toll Alerting" business. I realize >> some people love it, but I always thought it was silly, and now >> that long-distance is down in the 0-4 cents per minute range, is >> it really that big of a deal? >> >> Correct me if I'm wrong, but wasn't 1+ dialing introduced to >> disambiguate out-of-npa calls when generalized (from NNX to NXX) >> office codes were first assigned back in the early 70s? Back when NPAs >> were all N0/1X and office codes were all NNX, there were no O.C./npa >> collisions, so all calls could be dialed as 7D or 10D without >> ambiguity. So the idea of 1+ = toll alerting seems like an artifact, >> not something by design. >> >> Living in Los Angeles, where area codes are quite numerous, people >> have generally started just dialing 1+10D for all calls. Given that, >> it seems to me that the 1 is now just anachronistic baggage, so for >> my systems, I've dropped it. I just use 10D for all calls. You >> can use 7D but there's a timeout. >> >> The telcos still require the 1, however. >> >> Also, in 310, which is scheduled for an overlay, 1+10D is now >> mandatory on *all* calls. I understand the rationale here is that >> since the CLECs will be the ones primarily getting assignments in >> the new area code, the CPUC wanted to level they playing field by >> requiring 11D, thereby eliminating the preferability of 310 office >> codes. >> >> But anyway, meh on 1+ toll alerting, I say. From mark at rudholm.com Fri Jun 1 12:04:02 2007 From: mark at rudholm.com (Mark Rudholm) Date: Fri, 01 Jun 2007 10:04:02 -0700 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <46604791.1080809@stromberg-carlson.org> References: <46603ED6.9050500@rudholm.com> <46604791.1080809@stromberg-carlson.org> Message-ID: <46605182.6010308@rudholm.com> John Novack wrote: > > > Mark Rudholm wrote: >> Maybe I have a southern California bias or something, but I never >> really understood the whole "1+ Toll Alerting" business. I realize >> some people love it, but I always thought it was silly, > Historically, it was pretty much a requirement in Step offices to reach > the toll operator or for DDD. Some locales with party lines would dial > 112x with the X designating the party on a given line > In the larger Bell cities ( Except LA for other historical reasons ) > there was no 1+ dialing in the EM days > 211 would get one to the toll operator, then later dialing 10 digits > when the NPA was in the form of N0/1X would get directed properly > Things changed once we enter the electronic era and CLECs and NPA's with > the form NXX > So, NO, it wasn't/isn't silly Well, I think it's silly *now*. Maybe I'm terribly wealthy or something, but really, a couple cents is down in the noise. I think I misplace more money that that every month. And when does 1+ toll alerting kick in? Zone 3? beyond Zone 3? inter-LATA? > LA was, in the EM era, all or almost all step, Historically due to the > merging and buyout of independents, then I suspect due to the frugal > nature of the PUC regarding Pacific Bell. At one point AT&T seriously > wanted to rid itself of that unit due to the "frugal" nature of the > California PUC. See "The Fall of the Bell System" for much more > information on the corporate side of things. >> Given that,it seems to me that the 1 is now just anachronistic >> baggage, so for my systems, I've dropped it. > Yet several state PUC insist on dictating that it be used for toll, or > ALL calls beyond 7 digits. My telephonic memory starts around 1976, at that point, we had just been cut over to a 1A. The CO also had step and xbar, but I was lucky. >> I just use 10D for all calls. You can use 7D but there's a timeout. >> >> The telcos still require the 1, however. >> > In many cases, this is dictated to them by the PUC, or whatever it is > called in a given state. Fortunately, MarkTel doesn't fall under the jurisdiction of the CPUC :) > Part of the current insanity. In MO, for example, the PUC dictated where > a split was to be located, rather than growth patterns of the > population. I believe the only thing that saved SBC was the decline in > lines due to HSIA and release of 2nd lines along with the decline in FAX > usage. Yeah, and going to 1000 block number assignments helped provide relief for 310 as well. The overlay for it was originally scheduled for 1999. > IMHO, once the US was all electronic, there should NEVER have been > another split. Overlays cost everyone less, even though they require 10 > digit dialing in many cases. Intelligence in the switch could be used, > but it is much easier for the user to just dial 10 digits for all local > calls I'm a huge proponent of overlays, particularly in urban areas. The phone number my parents had since 1968 was recently assigned to a law firm downtown. :( >> Also, in 310, which is scheduled for an overlay, 1+10D is now >> mandatory on *all* calls. I understand the rationale here is that >> since the CLECs will be the ones primarily getting assignments in the >> new area code, the CPUC wanted to level they playing field by >> requiring 11D, thereby eliminating the preferability of 310 office codes. >> > That makes little sense. 10 digit local calls, 11 digit toll calls is > really the best solution. It helps the user determine toll vs local and > simplifys switch translations. Yeah, I had to train my customers (my parents and a couple of friends) to use 10D on all calls. It took them a while to give up the 1+ habit, but eventually they got used to and liked it. > Right now anyone traveling and not in the know can easily become > frustrated. > Add to that the cell phone dialing requirements > My carrier requires 10 digits for ALL calls since my home LATA has an > overlay and requires it, adds the 1 towards the network and calls go > through. Since all my cell calls are billed the same, there is no toll > vs local issue or time of day issue, calls to other cells in the same > carrier are free, so it is simply a minute issue . > The landline carriers and PUC's could learn from that. My cellphone is in 310, so 10D or 11D is required on all calls I have numbers in my cellphone's phone book programmed as "+1 NPA NXX XXXX" so that when I travel internationally, it just works. From ka2wft at arrl.net Fri Jun 1 12:06:55 2007 From: ka2wft at arrl.net (Doug Alderdice) Date: Fri, 01 Jun 2007 13:06:55 -0400 Subject: [VoIP] Numbering - existing use of 311 In-Reply-To: <46604791.1080809@stromberg-carlson.org> References: <46603ED6.9050500@rudholm.com> <46603ED6.9050500@rudholm.com> Message-ID: <5.1.0.14.2.20070601125934.00bb79d8@incoming.verizon.net> At 6/1/2007 12:21 PM -0400, John Novack wrote: >That makes little sense. 10 digit local calls, 11 digit toll calls is >really the best solution. It helps the user determine toll vs local and >simplifys switch translations. >Right now anyone traveling and not in the know can easily become frustrated. >Add to that the cell phone dialing requirements I experienced that first-hand this week. My uncle and his 13-yr-old grandson were visiting from Texas this week. I was showing the 13-yr-old my step switch (which he thought was really cool, and he could actually operate a rotary dial phone w/o instruction!) and after learning he could dial out to the PSTN from it, immediately tried to phone his folks back home through it. Except that being from the Dallas/Ft Worth Metroplex and their all-over 10D dialing, he was trying to phone home w/o the leading 1+ and he couldn't figure out why he couldn't get through. Once I told him that he had to add the 1+ the call, of course, went through. Doug. From hockd at dteenergy.com Fri Jun 1 12:16:36 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 13:16:36 -0400 Subject: [VoIP] Cisco 3810 Message-ID: Thanks i'll take a look and have to acquire it no doubt. Thank you for the input. Dennis -----voip-bounces at ckts.info wrote: ----- To: "Voice Over IP Tandem for Analog Switches" From: "Rusty Dekema" Sent by: voip-bounces at ckts.info Date: 06/01/2007 10:26AM Subject: Re: [VoIP] Cisco 3810 On 6/1/07, Dennis D Hock wrote: > John, that is to say I don't have it??? Any idea the cost?? ballpark > > Thanks, > > Dennis H Looks like about $30 on eBay. I forget exactly what kind of memory is needed for the MC3810, I think it's a parity SIMM. However, if you search for mc3810 on eBay, there are a ton of listings of "64M Memory Module for MC3810" for about $30 each. I'm pretty sure you need the 64M of RAM to run any version of IOS that supports SIP on the 3810, but I could be wrong about that. Rusty _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 12:18:29 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 13:18:29 -0400 Subject: [VoIP] Cisco 3810 Message-ID: I believe I have all the parts and ports I require only the memory is in querstion maybe. Dennis H. -----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: 06/01/2007 11:14AM Subject: Re: [VoIP] Cisco 3810 I am also a LITTLE confused about all of this The unit I ( sometimes ) have working has 2 replaceable memory modules One is marked SMART and then a long part number Is this the flash memory? The NVram? Another, shorter module is marked 64MB sync 125Mhz This is the main memory? Clearly one needs the 16 meg flash as the SIP image is 13 meg All old files have to be offloaded and deleted before the SIP image can be loaded There are often 3810's that show up on eBay, but not all will allow this configuration Not all have all the cards needed either, and purchasing the AVM daughterboard alone is $60, then up to 6 modules. Finding one properly equipped can often be the lowest cost. John Novack Rusty Dekema wrote: > On 6/1/07, Dennis D Hock wrote: > >> John, that is to say I don't have it??? Any idea the cost?? ballpark >> >> Thanks, >> >> Dennis H >> > > Looks like about $30 on eBay. I forget exactly what kind of memory is > needed for the MC3810, I think it's a parity SIMM. However, if you > search for mc3810 on eBay, there are a ton of listings of "64M Memory > Module for MC3810" for about $30 each. > > I'm pretty sure you need the 64M of RAM to run any version of IOS that > supports SIP on the 3810, but I could be wrong about that. > > > Rusty _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From jjones3601 at yahoo.com Fri Jun 1 13:39:36 2007 From: jjones3601 at yahoo.com (john jones) Date: Fri, 1 Jun 2007 11:39:36 -0700 (PDT) Subject: [VoIP] Cisco 3810 Message-ID: <20070601183936.27063.qmail@web34303.mail.mud.yahoo.com> The DRAM is usually provided with 2 slots, so Cisco can sell you a base unit with a respectable list price, and then gouge the hell of of you for memory upgrades. However, in your case, the shorter module is the DRAM. Is there only 1 short slot? John ----- Original Message ---- From: John Novack To: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 11:14:57 AM Subject: Re: [VoIP] Cisco 3810 I am also a LITTLE confused about all of this The unit I ( sometimes ) have working has 2 replaceable memory modules One is marked SMART and then a long part number Is this the flash memory? The NVram? Another, shorter module is marked 64MB sync 125Mhz This is the main memory? Clearly one needs the 16 meg flash as the SIP image is 13 meg All old files have to be offloaded and deleted before the SIP image can be loaded There are often 3810's that show up on eBay, but not all will allow this configuration Not all have all the cards needed either, and purchasing the AVM daughterboard alone is $60, then up to 6 modules. Finding one properly equipped can often be the lowest cost. John Novack Rusty Dekema wrote: > On 6/1/07, Dennis D Hock wrote: > >> John, that is to say I don't have it??? Any idea the cost?? ballpark >> >> Thanks, >> >> Dennis H >> > > Looks like about $30 on eBay. I forget exactly what kind of memory is > needed for the MC3810, I think it's a parity SIMM. However, if you > search for mc3810 on eBay, there are a ton of listings of "64M Memory > Module for MC3810" for about $30 each. > > I'm pretty sure you need the 64M of RAM to run any version of IOS that > supports SIP on the 3810, but I could be wrong about that. > > > Rusty _______________________________________________ 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 Fri Jun 1 13:42:30 2007 From: jjones3601 at yahoo.com (john jones) Date: Fri, 1 Jun 2007 11:42:30 -0700 (PDT) Subject: [VoIP] Cisco 3810 Message-ID: <774310.53649.qm@web34307.mail.mud.yahoo.com> No, It just wasn't in the list of specs you sent. I think we've received 3810's from the same guy and they seem to be max'd out. Just power it up to see. Let me know if you need a console cable. John ----- Original Message ---- From: Dennis D Hock To: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 9:48:46 AM Subject: Re: [VoIP] Cisco 3810 John, that is to say I don't have it??? Any idea the cost?? ballpark Thanks, Dennis H -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: john jones Sent by: voip-bounces at ckts.info Date: 06/01/2007 08:37AM cc: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 You also want to make sure you get 64M of DRAM, or factor in the cost of an upgrade if it isn't equipped that way from the beginning. John ----- Original Message ---- From: Dennis D Hock To: Voice Over IP Tandem for Analog Switches Cc: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 5:01:44 AM Subject: Re: [VoIP] Cisco 3810 Lee and others. the person I purchased the 3810 from was Chris at West tech recyclers out west. He is at chrisk at westechrecyclers.com. A short summary of the unit shows it has 1 serial interface, I channelized T1/PRI port, 256K non volatile config memory, 16384K bytes of processor board Sys flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 serial (sync/async) network interface. It also has a combo Analog Voice Module containign 4 FXS and 2 FXO. These folks were very easy to do business with. I purchased two units ( one for me, one for a fellow at work). The only problem we had so far was they seemed to have lost the order in their shipping dept. It was however very quickly made right when I hadn't received the units and inquired. It sounded like they had this package and had lost the address it was to be sent to. I would have no problem using them again. At the time he had a selection of these and of course other units to choose from. Hope this helps all. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches , jnovack at stromberg-carlson.org From: john jones Sent by: voip-bounces at ckts.info Date: 05/31/2007 07:39PM Subject: Re: [VoIP] Cisco 3810 The FXO port supports DTMF, MF, and pulse dialing. On one of my 3810's, ( I have routers like you guys have phones) has a FXO port in slot 1/6. To change the out-pulsing mode, use the following command BatPhone(config-voiceport)#dial BatPhone(config-voiceport)#dial-type ? dtmf touch-tone dialer mf mf-tone dialer pulse pulse dialer Just pick the one you want. The E&M interface is exactly the same. The FXS port supports DTMF and pulse dialing. I've never tried MF on an FXS port and I'm not going to try it now! John ----- Original Message ---- From: Lee Spenadel To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Sent: Thursday, May 31, 2007 7:14:56 AM Subject: Re: [VoIP] Cisco 3810 So was I. How much was it? I'm wondering if that would be a nice device to use for FXO in lieu of the Adtran given the outpulsing problem toward my SxS. L -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Thursday, May 31, 2007 12:10 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 I see Russ Price is awake this late to enjoy my Cisco! JN John Novack wrote: > On a lighter note, with John Jones expert help, I have been working a > little with the Cisco 3810 > Reloaded with SIP software, the box can be equipped with 6 ports, in any > mix, of FXS, FXO or E&M circuits. > Paul Wills is using one for his switch and ANI. > I have mine on line right now set up for several foreign Ringback tones. > Though it may not be available at all times, give it a call at 666-3101 > through 3104 for Cisco's electronic version of some other than North > American precise tones. UK, RU, EG and JP > No phones connected, so one can call at odd hours. > > 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/ _______________________________________________ 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/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voip Project Web Page: http://www.ckts.info/ From hockd at dteenergy.com Fri Jun 1 13:48:46 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 14:48:46 -0400 Subject: [VoIP] Cisco 3810 Message-ID: OK, don't need the console cable as I already have one. Are there any parcticular settings to setup the console port (probably should look in the book). Dennis -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: john jones Sent by: voip-bounces at ckts.info Date: 06/01/2007 02:42PM Subject: Re: [VoIP] Cisco 3810 No, It just wasn't in the list of specs you sent. I think we've received 3810's from the same guy and they seem to be max'd out. Just power it up to see. Let me know if you need a console cable. John ----- Original Message ---- From: Dennis D Hock To: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 9:48:46 AM Subject: Re: [VoIP] Cisco 3810 John, that is to say I don't have it??? Any idea the cost?? ballpark Thanks, Dennis H -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches From: john jones Sent by: voip-bounces at ckts.info Date: 06/01/2007 08:37AM cc: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 You also want to make sure you get 64M of DRAM, or factor in the cost of an upgrade if it isn't equipped that way from the beginning. John ----- Original Message ---- From: Dennis D Hock To: Voice Over IP Tandem for Analog Switches Cc: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 5:01:44 AM Subject: Re: [VoIP] Cisco 3810 Lee and others. the person I purchased the 3810 from was Chris at West tech recyclers out west. He is at chrisk at westechrecyclers.com. A short summary of the unit shows it has 1 serial interface, I channelized T1/PRI port, 256K non volatile config memory, 16384K bytes of processor board Sys flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 serial (sync/async) network interface. It also has a combo Analog Voice Module containign 4 FXS and 2 FXO. These folks were very easy to do business with. I purchased two units ( one for me, one for a fellow at work). The only problem we had so far was they seemed to have lost the order in their shipping dept. It was however very quickly made right when I hadn't received the units and inquired. It sounded like they had this package and had lost the address it was to be sent to. I would have no problem using them again. At the time he had a selection of these and of course other units to choose from. Hope this helps all. Dennis Hock -----voip-bounces at ckts.info wrote: ----- To: Voice Over IP Tandem for Analog Switches , jnovack at stromberg-carlson.org From: john jones Sent by: voip-bounces at ckts.info Date: 05/31/2007 07:39PM Subject: Re: [VoIP] Cisco 3810 The FXO port supports DTMF, MF, and pulse dialing. On one of my 3810's, ( I have routers like you guys have phones) has a FXO port in slot 1/6. To change the out-pulsing mode, use the following command BatPhone(config-voiceport)#dial BatPhone(config-voiceport)#dial-type ? dtmf touch-tone dialer mf mf-tone dialer pulse pulse dialer Just pick the one you want. The E&M interface is exactly the same. The FXS port supports DTMF and pulse dialing. I've never tried MF on an FXS port and I'm not going to try it now! John ----- Original Message ---- From: Lee Spenadel To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches Sent: Thursday, May 31, 2007 7:14:56 AM Subject: Re: [VoIP] Cisco 3810 So was I. How much was it? I'm wondering if that would be a nice device to use for FXO in lieu of the Adtran given the outpulsing problem toward my SxS. L -----Original Message----- From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of John Novack Sent: Thursday, May 31, 2007 12:10 AM To: Voice Over IP Tandem for Analog Switches Subject: Re: [VoIP] Cisco 3810 I see Russ Price is awake this late to enjoy my Cisco! JN John Novack wrote: > On a lighter note, with John Jones expert help, I have been working a > little with the Cisco 3810 > Reloaded with SIP software, the box can be equipped with 6 ports, in any > mix, of FXS, FXO or E&M circuits. > Paul Wills is using one for his switch and ANI. > I have mine on line right now set up for several foreign Ringback tones. > Though it may not be available at all times, give it a call at 666-3101 > through 3104 for Cisco's electronic version of some other than North > American precise tones. UK, RU, EG and JP > No phones connected, so one can call at odd hours. > > 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/ _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProjectWebPage: 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/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/ _______________________________________________ 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 Fri Jun 1 13:52:04 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Fri, 01 Jun 2007 14:52:04 -0400 Subject: [VoIP] Cisco 3810 In-Reply-To: <20070601183936.27063.qmail@web34303.mail.mud.yahoo.com> References: <20070601183936.27063.qmail@web34303.mail.mud.yahoo.com> Message-ID: <46606AD4.4000109@stromberg-carlson.org> john jones wrote: > The DRAM is usually provided with 2 slots, so Cisco can sell you a base unit with a respectable list price, and then gouge the hell of of you for memory upgrades. However, in your case, the shorter module is the DRAM. Is there only 1 short slot? > > John > > Correct. One short, one longer. In troubleshooting the other unit, I swapped these modules, which seemed to put the working one into a tailspin of sorts I can no longer call out from the FXS ports, so when I can, I will revisit that, as it WAS working! MORE FUN than any one person should have - NOT! John Novack > ----- Original Message ---- > From: John Novack > To: Voice Over IP Tandem for Analog Switches > Sent: Friday, June 1, 2007 11:14:57 AM > Subject: Re: [VoIP] Cisco 3810 > > I am also a LITTLE confused about all of this > The unit I ( sometimes ) have working has 2 replaceable memory modules > One is marked SMART and then a long part number > Is this the flash memory? The NVram? > Another, shorter module is marked 64MB sync 125Mhz > This is the main memory? > > Clearly one needs the 16 meg flash as the SIP image is 13 meg > All old files have to be offloaded and deleted before the SIP image can > be loaded > > There are often 3810's that show up on eBay, but not all will allow this > configuration > Not all have all the cards needed either, and purchasing the AVM > daughterboard alone is $60, then up to 6 modules. > Finding one properly equipped can often be the lowest cost. > > John Novack > > Rusty Dekema wrote: > >> On 6/1/07, Dennis D Hock wrote: >> >> >>> John, that is to say I don't have it??? Any idea the cost?? ballpark >>> >>> Thanks, >>> >>> Dennis H >>> >>> >> Looks like about $30 on eBay. I forget exactly what kind of memory is >> needed for the MC3810, I think it's a parity SIMM. However, if you >> search for mc3810 on eBay, there are a ton of listings of "64M Memory >> Module for MC3810" for about $30 each. >> >> I'm pretty sure you need the 64M of RAM to run any version of IOS that >> supports SIP on the 3810, but I could be wrong about that. >> >> >> Rusty >> > _______________________________________________ > 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 Fri Jun 1 13:55:15 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Fri, 01 Jun 2007 14:55:15 -0400 Subject: [VoIP] Cisco 3810 In-Reply-To: References: Message-ID: <46606B93.4010304@stromberg-carlson.org> I used hyperterminal from a windows machine set to 9600,n,8,1 and NO flow control Once you get the Ethernet port set up you can telnet to the box. John Jones is the highest authority on this though. From my experiences, cut and pasting of the commands doesn't work well, due to lack of any flow control. That seems to be the way Cisco set up the console port! John Novack Dennis D Hock wrote: > OK, don't need the console cable as I already have one. Are there any > parcticular settings to setup the console port (probably should look in the > book). > > Dennis > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: john jones > Sent by: voip-bounces at ckts.info > Date: 06/01/2007 02:42PM > Subject: Re: [VoIP] Cisco 3810 > > No, It just wasn't in the list of specs you sent. I think we've received > 3810's from the same guy and they seem to be max'd out. > > > Just power it up to see. > > Let me know if you need a console cable. > > John > > ----- Original Message ---- > From: Dennis D Hock > To: Voice Over IP Tandem for Analog Switches > Sent: Friday, June 1, 2007 9:48:46 AM > Subject: Re: [VoIP] Cisco 3810 > > John, that is to say I don't have it??? Any idea the cost?? ballpark > > Thanks, > > Dennis H > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: john jones > Sent by: voip-bounces at ckts.info > Date: 06/01/2007 08:37AM > cc: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Cisco 3810 > > You also want to make sure you get 64M of DRAM, or factor in the cost of an > upgrade if it isn't equipped that way from the beginning. > > John > > ----- Original Message ---- > From: Dennis D Hock > To: Voice Over IP Tandem for Analog Switches > Cc: Voice Over IP Tandem for Analog Switches > Sent: Friday, June 1, 2007 5:01:44 AM > Subject: Re: [VoIP] Cisco 3810 > > > Lee and others. the person I purchased the 3810 from was Chris at West > tech recyclers out west. He is at chrisk at westechrecyclers.com. A short > summary of the unit shows it has 1 serial interface, I channelized T1/PRI > port, 256K non volatile config memory, 16384K bytes of processor board Sys > flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 > serial (sync/async) network interface. It also has a combo Analog Voice > Module containign 4 FXS and 2 FXO. > > These folks were very easy to do business with. I purchased two units ( > one for me, one for a fellow at work). The only problem we had so far was > they seemed to have lost the order in their shipping dept. It was however > very quickly made right when I hadn't received the units and inquired. It > sounded like they had this package and had lost the address it was to be > sent to. I would have no problem using them again. > > At the time he had a selection of these and of course other units to choose > from. > > Hope this helps all. > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches , > jnovack at stromberg-carlson.org > From: john jones > Sent by: voip-bounces at ckts.info > Date: 05/31/2007 07:39PM > Subject: Re: [VoIP] Cisco 3810 > > The FXO port supports DTMF, MF, and pulse dialing. > > On one of my 3810's, ( I have routers like you guys have phones) has a FXO > port in slot 1/6. > > To change the out-pulsing mode, use the following command > > BatPhone(config-voiceport)#dial > BatPhone(config-voiceport)#dial-type ? > dtmf touch-tone dialer > mf mf-tone dialer > pulse pulse dialer > > > > Just pick the one you want. > > The E&M interface is exactly the same. > > The FXS port supports DTMF and pulse dialing. I've never tried MF on an > FXS port and I'm not going to try it now! > > John > > > ----- Original Message ---- > From: Lee Spenadel > To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches > > Sent: Thursday, May 31, 2007 7:14:56 AM > Subject: Re: [VoIP] Cisco 3810 > > So was I. How much was it? I'm wondering if that would be a nice device > to > use for FXO in lieu of the Adtran given the outpulsing problem toward my > SxS. > > L > > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of > John Novack > Sent: Thursday, May 31, 2007 12:10 AM > To: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Cisco 3810 > > I see Russ Price is awake this late to enjoy my Cisco! > > JN > > > John Novack wrote: > >> On a lighter note, with John Jones expert help, I have been working a >> little with the Cisco 3810 >> Reloaded with SIP software, the box can be equipped with 6 ports, in any >> mix, of FXS, FXO or E&M circuits. >> Paul Wills is using one for his switch and ANI. >> I have mine on line right now set up for several foreign Ringback tones. >> Though it may not be available at all times, give it a call at 666-3101 >> through 3104 for Cisco's electronic version of some other than North >> American precise tones. UK, RU, EG and JP >> No phones connected, so one can call at odd hours. >> >> 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/ > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voipProjectWebPage: > 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/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/ > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voipProject Web Page: > http://www.ckts.info/ > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voip > Project Web Page: http://www.ckts.info/ > > From hockd at dteenergy.com Fri Jun 1 14:13:26 2007 From: hockd at dteenergy.com (Dennis D Hock) Date: Fri, 1 Jun 2007 15:13:26 -0400 Subject: [VoIP] Cisco 3810 Message-ID: Got it thanks. I figured JJ was the word as we had some discussions before I opted to invest in this one. It was his patient tutelage which convinced me. Dennis -----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: 06/01/2007 02:55PM Subject: Re: [VoIP] Cisco 3810 I used hyperterminal from a windows machine set to 9600,n,8,1 and NO flow control Once you get the Ethernet port set up you can telnet to the box. John Jones is the highest authority on this though. >From my experiences, cut and pasting of the commands doesn't work well, due to lack of any flow control. That seems to be the way Cisco set up the console port! John Novack Dennis D Hock wrote: > OK, don't need the console cable as I already have one. Are there any > parcticular settings to setup the console port (probably should look in the > book). > > Dennis > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: john jones > Sent by: voip-bounces at ckts.info > Date: 06/01/2007 02:42PM > Subject: Re: [VoIP] Cisco 3810 > > No, It just wasn't in the list of specs you sent. I think we've received > 3810's from the same guy and they seem to be max'd out. > > > Just power it up to see. > > Let me know if you need a console cable. > > John > > ----- Original Message ---- > From: Dennis D Hock > To: Voice Over IP Tandem for Analog Switches > Sent: Friday, June 1, 2007 9:48:46 AM > Subject: Re: [VoIP] Cisco 3810 > > John, that is to say I don't have it??? Any idea the cost?? ballpark > > Thanks, > > Dennis H > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: john jones > Sent by: voip-bounces at ckts.info > Date: 06/01/2007 08:37AM > cc: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Cisco 3810 > > You also want to make sure you get 64M of DRAM, or factor in the cost of an > upgrade if it isn't equipped that way from the beginning. > > John > > ----- Original Message ---- > From: Dennis D Hock > To: Voice Over IP Tandem for Analog Switches > Cc: Voice Over IP Tandem for Analog Switches > Sent: Friday, June 1, 2007 5:01:44 AM > Subject: Re: [VoIP] Cisco 3810 > > > Lee and others. the person I purchased the 3810 from was Chris at West > tech recyclers out west. He is at chrisk at westechrecyclers.com. A short > summary of the unit shows it has 1 serial interface, I channelized T1/PRI > port, 256K non volatile config memory, 16384K bytes of processor board Sys > flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 > serial (sync/async) network interface. It also has a combo Analog Voice > Module containign 4 FXS and 2 FXO. > > These folks were very easy to do business with. I purchased two units ( > one for me, one for a fellow at work). The only problem we had so far was > they seemed to have lost the order in their shipping dept. It was however > very quickly made right when I hadn't received the units and inquired. It > sounded like they had this package and had lost the address it was to be > sent to. I would have no problem using them again. > > At the time he had a selection of these and of course other units to choose > from. > > Hope this helps all. > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches , > jnovack at stromberg-carlson.org > From: john jones > Sent by: voip-bounces at ckts.info > Date: 05/31/2007 07:39PM > Subject: Re: [VoIP] Cisco 3810 > > The FXO port supports DTMF, MF, and pulse dialing. > > On one of my 3810's, ( I have routers like you guys have phones) has a FXO > port in slot 1/6. > > To change the out-pulsing mode, use the following command > > BatPhone(config-voiceport)#dial > BatPhone(config-voiceport)#dial-type ? > dtmf touch-tone dialer > mf mf-tone dialer > pulse pulse dialer > > > > Just pick the one you want. > > The E&M interface is exactly the same. > > The FXS port supports DTMF and pulse dialing. I've never tried MF on an > FXS port and I'm not going to try it now! > > John > > > ----- Original Message ---- > From: Lee Spenadel > To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches > > Sent: Thursday, May 31, 2007 7:14:56 AM > Subject: Re: [VoIP] Cisco 3810 > > So was I. How much was it? I'm wondering if that would be a nice device > to > use for FXO in lieu of the Adtran given the outpulsing problem toward my > SxS. > > L > > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of > John Novack > Sent: Thursday, May 31, 2007 12:10 AM > To: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Cisco 3810 > > I see Russ Price is awake this late to enjoy my Cisco! > > JN > > > John Novack wrote: > >> On a lighter note, with John Jones expert help, I have been working a >> little with the Cisco 3810 >> Reloaded with SIP software, the box can be equipped with 6 ports, in any >> mix, of FXS, FXO or E&M circuits. >> Paul Wills is using one for his switch and ANI. >> I have mine on line right now set up for several foreign Ringback tones. >> Though it may not be available at all times, give it a call at 666-3101 >> through 3104 for Cisco's electronic version of some other than North >> American precise tones. UK, RU, EG and JP >> No phones connected, so one can call at odd hours. >> >> 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/ > > > > _______________________________________________ > VoIP mailing list > VoIP at ckts.info > http://lists.ckts.info/mailman/listinfo/voipProjectWebPage: > 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/voipProjectWebPage: > 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/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/ > > _______________________________________________ VoIP mailing list VoIP at ckts.info http://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/ From lee at spenadel.com Fri Jun 1 14:24:39 2007 From: lee at spenadel.com (Lee Spenadel) Date: Fri, 1 Jun 2007 15:24:39 -0400 Subject: [VoIP] You're Welcome Message-ID: <00dd01c7a482$7fa51ee0$7eef5ca0$@com> If anyone is visiting Cape Cod this summer, please contact me if you would like to see my telephone collection/switching systems. I'd love to show it off. Regards Lee If your car could travel at the speed of light, would your headlights work? From jnovack at stromberg-carlson.org Fri Jun 1 14:29:00 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Fri, 01 Jun 2007 14:29:00 -0500 (CDT) Subject: [VoIP] Cisco 3810, Linksys Router and Call Data Records Message-ID: <31580815.1488161180726140844.JavaMail.root@vms064.mailsrvcs.net> Here's a variation on a theme: My 3810 has been working for a couple of months now and I am very impressed. The sound quality is much better than anything obtained with the "PC based" FXS and FXO boards. Outbound and inbound signalling have been flawless and it sits nicely in the equipment rack under the polar relay test panel. Once I got the 3810 running, there was absolutely no need to host any IO in the Asterisk processor. Having found out about an alternative Asterisk platform from Mr. Jones, I figured it was time to get rid of the PC completely and conserve power. The Linksys router (a WRTSL54GS) has just enough space to run a minimal version of Linux and Asterisk. The best part about the WRTSL54GS is that it has a USB port which allows for the use of a flash memory to hold all the sound files and logs. (With no moving parts!) I believe the entire power requirement is a measly 5 watts. To address several operating problems, I upgraded the Linux to the latest version they had for the router as well as Asterisk to 1.4. Unfortunately, 1.4 just doesn't stream audio data as well as a larger processor so I eventually went back to 1.2 with better results. (The operating problems eventually turned out to be related to other issues, since resolved, that had nothing to do with the Linux or Asterisk versions.) As a basic switch, it works fine. There is no voice mail to speak of and the lack of an internal Zaptel timer limits a few other functions that I don't use. This all leads to one question: There is absolutely no more room to install anything so I can't install a PHP server or any database software. All the CDR viewers I find need PHP and MySQL installed on the server to work. Does anyone know of something that would run on a Windows machine and simply read the shared CDR file that "lives" on the Asterisk box? PDW From ikj1234i at yahoo.com Fri Jun 1 15:10:47 2007 From: ikj1234i at yahoo.com (ikjtel) Date: Fri, 1 Jun 2007 13:10:47 -0700 (PDT) Subject: [VoIP] Adtran outpulsing In-Reply-To: <465FA2DD.2030501@stromberg-carlson.org> Message-ID: <20070601201047.68432.qmail@web51603.mail.re2.yahoo.com> --- John Novack wrote: > I have a TDM400 still in my system, ( Max, you DID > mean a TDM400 and > not a Digium T1 card, didn't you?? ) Yep. My guess is that you won't have the erratic pulsing with the TDM400 - but it would be good to know for sure so as to eliminate one more variable... Max ____________________________________________________________________________________ Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. http://mobile.yahoo.com/go?refer=1GNXIC From mark at rudholm.com Fri Jun 1 15:22:13 2007 From: mark at rudholm.com (Mark Rudholm) Date: Fri, 01 Jun 2007 13:22:13 -0700 Subject: [VoIP] Adtran outpulsing In-Reply-To: <20070601201047.68432.qmail@web51603.mail.re2.yahoo.com> References: <20070601201047.68432.qmail@web51603.mail.re2.yahoo.com> Message-ID: <46607FF5.7070606@rudholm.com> ikjtel wrote: > --- John Novack wrote: > >> I have a TDM400 still in my system, ( Max, you DID >> mean a TDM400 and >> not a Digium T1 card, didn't you?? ) > > Yep. My guess is that you won't have the erratic > pulsing with the TDM400 - but it would be good to know > for sure so as to eliminate one more variable... That reminds me, I recently upgraded from a TDM400 (1 FXO + 3 FXS) to a TDM800 (1 FXO + 5 FXS). If anyone wants the displaced hardware; TDM400 + 2 FXS modules, let me know. I'll sell it cheap, otherwise it's going up on ebay starting around 190$ (which seems to be the going price). From jnovack at stromberg-carlson.org Fri Jun 1 15:26:26 2007 From: jnovack at stromberg-carlson.org (John Novack) Date: Fri, 01 Jun 2007 16:26:26 -0400 Subject: [VoIP] Cisco 3810, Linksys Router and Call Data Records In-Reply-To: <31580815.1488161180726140844.JavaMail.root@vms064.mailsrvcs.net> References: <31580815.1488161180726140844.JavaMail.root@vms064.mailsrvcs.net> Message-ID: <466080F2.3050605@stromberg-carlson.org> Curious, since Paul Will sent this, not me!! Has he broken into my Win2K machine?? The CDR that NORMAL Asterisk writes can be read by Excell. Is there even a CDR written with your White Russian version?? NOT PAUL WILLS! John Novack wrote: > Here's a variation on a theme: > > My 3810 has been working for a couple of months now and I am very impressed. The sound quality is much better than anything obtained with the "PC based" FXS and FXO boards. Outbound and inbound signalling have been flawless and it sits nicely in the equipment rack under the polar relay test panel. > > Once I got the 3810 running, there was absolutely no need to host any IO in the Asterisk processor. Having found out about an alternative Asterisk platform from Mr. Jones, I figured it was time to get rid of the PC completely and conserve power. The Linksys router (a WRTSL54GS) has just enough space to run a minimal version of Linux and Asterisk. The best part about the WRTSL54GS is that it has a USB port which allows for the use of a flash memory to hold all the sound files and logs. (With no moving parts!) I believe the entire power requirement is a measly 5 watts. > > To address several operating problems, I upgraded the Linux to the latest version they had for the router as well as Asterisk to 1.4. Unfortunately, 1.4 just doesn't stream audio data as well as a larger processor so I eventually went back to 1.2 with better results. (The operating problems eventually turned out to be related to other issues, since resolved, that had nothing to do with the Linux or Asterisk versions.) As a basic switch, it works fine. There is no voice mail to speak of and the lack of an internal Zaptel timer limits a few other functions that I don't use. > > This all leads to one question: > > There is absolutely no more room to install anything so I can't install a PHP server or any database software. All the CDR viewers I find need PHP and MySQL installed on the server to work. Does anyone know of something that would run on a Windows machine and simply read the shared CDR file that "lives" on the Asterisk box? > > PDW > > > _______________________________________________ > 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 Fri Jun 1 15:42:32 2007 From: pdwills at cedarknolltelephone.com (Paul Wills) Date: Fri, 01 Jun 2007 16:42:32 -0400 Subject: [VoIP] Cisco 3810, Linksys Router and Call Data Records References: <31580815.1488161180726140844.JavaMail.root@vms064.mailsrvcs.net> <466080F2.3050605@stromberg-carlson.org> Message-ID: <001401c7a48e$61f69610$0301a8c0@Main> I sent that last one via the Verizon web access replying to a message that John originally wrote. I have no idea how the Verizon webmail site can swap a reply address like that. In response to John's comments: 1) The CDR I get is very readable by Excell. I am just looking for asthetics. Perhaps an Excell macro is all I need. 2) The CDR is the same as any other Asterisk. That's one of the things that one should move to the USB flash drive to save space. PDW ----- Original Message ----- From: "John Novack" To: ; "Voice Over IP Tandem for Analog Switches" Sent: Friday, June 01, 2007 4:26 PM Subject: Re: [VoIP] Cisco 3810, Linksys Router and Call Data Records > Curious, since Paul Will sent this, not me!! > Has he broken into my Win2K machine?? > > The CDR that NORMAL Asterisk writes can be read by Excell. > Is there even a CDR written with your White Russian version?? > > NOT PAUL WILLS! > > John Novack wrote: >> Here's a variation on a theme: >> >> My 3810 has been working for a couple of months now and I am very >> impressed. The sound quality is much better than anything obtained with >> the "PC based" FXS and FXO boards. Outbound and inbound signalling have >> been flawless and it sits nicely in the equipment rack under the polar >> relay test panel. >> >> Once I got the 3810 running, there was absolutely no need to host any IO >> in the Asterisk processor. Having found out about an alternative >> Asterisk platform from Mr. Jones, I figured it was time to get rid of the >> PC completely and conserve power. The Linksys router (a WRTSL54GS) has >> just enough space to run a minimal version of Linux and Asterisk. The >> best part about the WRTSL54GS is that it has a USB port which allows for >> the use of a flash memory to hold all the sound files and logs. (With no >> moving parts!) I believe the entire power requirement is a measly 5 >> watts. >> >> To address several operating problems, I upgraded the Linux to the latest >> version they had for the router as well as Asterisk to 1.4. >> Unfortunately, 1.4 just doesn't stream audio data as well as a larger >> processor so I eventually went back to 1.2 with better results. (The >> operating problems eventually turned out to be related to other issues, >> since resolved, that had nothing to do with the Linux or Asterisk >> versions.) As a basic switch, it works fine. There is no voice mail to >> speak of and the lack of an internal Zaptel timer limits a few other >> functions that I don't use. >> >> This all leads to one question: >> >> There is absolutely no more room to install anything so I can't install a >> PHP server or any database software. All the CDR viewers I find need PHP >> and MySQL installed on the server to work. Does anyone know of something >> that would run on a Windows machine and simply read the shared CDR file >> that "lives" on the Asterisk box? >> >> PDW From jjones3601 at yahoo.com Fri Jun 1 16:55:34 2007 From: jjones3601 at yahoo.com (john jones) Date: Fri, 1 Jun 2007 14:55:34 -0700 (PDT) Subject: [VoIP] Cisco 3810 Message-ID: <221141.35344.qm@web34313.mail.mud.yahoo.com> Oh yeah, I forgot about flow control. I really need to put together a quick document on getting these things going. If you set up minimal commands via the async console port and then switch to telnet, flow control (and cut and paste) work much better. John ----- Original Message ---- From: John Novack To: Voice Over IP Tandem for Analog Switches Sent: Friday, June 1, 2007 2:55:15 PM Subject: Re: [VoIP] Cisco 3810 I used hyperterminal from a windows machine set to 9600,n,8,1 and NO flow control Once you get the Ethernet port set up you can telnet to the box. John Jones is the highest authority on this though. From my experiences, cut and pasting of the commands doesn't work well, due to lack of any flow control. That seems to be the way Cisco set up the console port! John Novack Dennis D Hock wrote: > OK, don't need the console cable as I already have one. Are there any > parcticular settings to setup the console port (probably should look in the > book). > > Dennis > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: john jones > Sent by: voip-bounces at ckts.info > Date: 06/01/2007 02:42PM > Subject: Re: [VoIP] Cisco 3810 > > No, It just wasn't in the list of specs you sent. I think we've received > 3810's from the same guy and they seem to be max'd out. > > > Just power it up to see. > > Let me know if you need a console cable. > > John > > ----- Original Message ---- > From: Dennis D Hock > To: Voice Over IP Tandem for Analog Switches > Sent: Friday, June 1, 2007 9:48:46 AM > Subject: Re: [VoIP] Cisco 3810 > > John, that is to say I don't have it??? Any idea the cost?? ballpark > > Thanks, > > Dennis H > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches > From: john jones > Sent by: voip-bounces at ckts.info > Date: 06/01/2007 08:37AM > cc: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Cisco 3810 > > You also want to make sure you get 64M of DRAM, or factor in the cost of an > upgrade if it isn't equipped that way from the beginning. > > John > > ----- Original Message ---- > From: Dennis D Hock > To: Voice Over IP Tandem for Analog Switches > Cc: Voice Over IP Tandem for Analog Switches > Sent: Friday, June 1, 2007 5:01:44 AM > Subject: Re: [VoIP] Cisco 3810 > > > Lee and others. the person I purchased the 3810 from was Chris at West > tech recyclers out west. He is at chrisk at westechrecyclers.com. A short > summary of the unit shows it has 1 serial interface, I channelized T1/PRI > port, 256K non volatile config memory, 16384K bytes of processor board Sys > flash (AMD29F016), 1 2DSP High perfoprmance Compression Module (v01.A0, 2 > serial (sync/async) network interface. It also has a combo Analog Voice > Module containign 4 FXS and 2 FXO. > > These folks were very easy to do business with. I purchased two units ( > one for me, one for a fellow at work). The only problem we had so far was > they seemed to have lost the order in their shipping dept. It was however > very quickly made right when I hadn't received the units and inquired. It > sounded like they had this package and had lost the address it was to be > sent to. I would have no problem using them again. > > At the time he had a selection of these and of course other units to choose > from. > > Hope this helps all. > > Dennis Hock > > -----voip-bounces at ckts.info wrote: ----- > > > To: Voice Over IP Tandem for Analog Switches , > jnovack at stromberg-carlson.org > From: john jones > Sent by: voip-bounces at ckts.info > Date: 05/31/2007 07:39PM > Subject: Re: [VoIP] Cisco 3810 > > The FXO port supports DTMF, MF, and pulse dialing. > > On one of my 3810's, ( I have routers like you guys have phones) has a FXO > port in slot 1/6. > > To change the out-pulsing mode, use the following command > > BatPhone(config-voiceport)#dial > BatPhone(config-voiceport)#dial-type ? > dtmf touch-tone dialer > mf mf-tone dialer > pulse pulse dialer > > > > Just pick the one you want. > > The E&M interface is exactly the same. > > The FXS port supports DTMF and pulse dialing. I've never tried MF on an > FXS port and I'm not going to try it now! > > John > > > ----- Original Message ---- > From: Lee Spenadel > To: jnovack at stromberg-carlson.org; Voice Over IP Tandem for Analog Switches > > Sent: Thursday, May 31, 2007 7:14:56 AM > Subject: Re: [VoIP] Cisco 3810 > > So was I. How much was it? I'm wondering if that would be a nice device > to > use for FXO in lieu of the Adtran given the outpulsing problem toward my > SxS. > > L > > -----Original Message----- > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of > John Novack > Sent: Thursday, May 31, 2007 12:10 AM > To: Voice Over IP Tandem for Analog Switches > Subject: Re: [VoIP] Cisco 3810 > > I see Russ Price is awake this late to enjoy my Cisco! > > JN > > > John Novack wrote: > >> On a lighter note, with John Jones expert help, I have been working a >> little with the Cisco 3810 >> Reloaded with SIP software, the box can be equipped with 6 ports, in any >> mix, of FXS, FXO or E&M circuits. >> Paul Wills is using one for his switch and ANI. >> I have mine on line right now set up for several foreign Ringback tones. >> Though it may not be available at all times, give it a call at 666-3101 >> through 3104 for Cisco's electronic version of some other than North >> American precise tones. UK, RU, EG and JP >> No phones connected, so one can call at odd hours. >> >> 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/ > > > > _________________________________________