Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
On Tue, 9 Sep 2003, Eric Wieling wrote: > It would have to do some kind of trascoding, Forgive my ignorance, but why? PSTN is delivering 8 bit 8 KHz ulaw samples. G711 is delivering 8 bit 8 KHz ulaw samples over SIP. Aren't the two data streams identical down to the bit level? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
On Tue, 9 Sep 2003, Eric Wieling wrote: > Transcoding would be required for access to ANY of the asterisk > sound files, voicemail and PSTN via Zap interfaces. If you are using G711 ulaw from the SIP phones, and that is what you are getting from the T1 PSTN link, would * have to transcode that? Is there more to it than digital to digital copy? Perhaps echo canceling? Can we also store sound files in ulaw? I know that takes more space, but perhaps it is less CPU work to move the bits around than to codec them. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940/7960 ethernet ports
On Mon, 8 Sep 2003, Travis Johnson wrote: > We are having a problem with Cisco 7940 and 7960 phones when > the PC is plugged into the 2nd ethernet port on the phone. It > will drop the PC's connection for about 30 seconds, then bring > the connection back up for about 30 seconds. It does this > continually regardless of how the port is configured on the > phone. > > Has anyone else seen this problem/issue? I've not seen this or used Cisco phones (yet), but I have witnessed a similar issue and it appears to be related to link negotiation of the ethernet switch to the PC. Try this (just as a test): use a long ethernet cable between the phone and the PC, say 50-100ft. If that "fixes" it, then the ethernet signal is "too strong". Sometimes, using really short cables and due to manufacturing tolerances, the signal can be too strong and the link will fail. This is a complex thing having to do with equalization, eye patterns, and possible high frequency noise on the wire. How long it takes to "fail" a link varies, but 30 seconds may be the time needed for the error threshold to be reached. The long cable filters the signal somewhat and lowers the amplitude. Another test: use a different NIC in the PC (as different as you can). Another test: force PC to be 10 Mbs link. Another test: look at PC NIC statistics and see if you are receiving bad frames (CRC, format error, etc). Ultimately, you may find that certain NICs with certain phones just do this and you have to find another brand/model NIC to make this happy. Disclaimer: everything in the email could be wrong. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk phone system plan - for review!
ice for certain employees to be able to have an IP phone at their house via cable modem. These phones would talk to the * box via the internet and they would become "extensions". I have not research the various issues of NAT, delay, bandwidth, etc. It would see we might want to choose something other than G.711 ulaw for these connections to get lower bandwidth. 11. I'd like to get a digital fax receive setup somewhere in this mess. Does the PM3 do it? Otherwise, a fax modem in the * box connected to one of the FXS lines out of the TA 750 seems doable but silly. Seems like we out to be able to decode the fax directly. Initial plan is to use existing fax machine on FXS port and hard copy output, but I hate fax spam. It's clear my plan is overkill for our immediate needs, but we expect to grow at this new location by perhaps a factor of 4. We can then enable 24 incoming lines on the T1, support essentially infinite modems, and support 24 FXS devices as designed. An additional Adtran TA 750 gets us another 24 FXS lines if we need them. If we need more, another TE410P card would expand us beyond reason. At some point, we may go IP dial tone, or perhaps just LD and toll free inbound. So, do this plan make sense? Where am I pushing the boundaries of the technology? Where will be my trouble points? Thanks for everyone's help! PS: I'm looking for a consultant to "hand hold" me if I get in trouble with config, setup, etc. I don't think any on site work will be needed and we can handle stuff like "check out the latest CVS and compile it". If you are in this line of work and really know your way around * and the equipment listed above, please send me a note with your areas of expertise, experience, and rates. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te410p with serial console fails with error:TE410P: Double/missed interrupt detected
On 1 Sep 2003, Klaus-Peter Junghanns wrote: > here is the URL for the netconsole patches: > http://www.kernel.org/pub/linux/people/mingo/netconsole-patches No work for me, instead: http://people.redhat.com/mingo/netconsole-patches/ Is that what you meant? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Difference between Cisco 7940/7940G, 7960/7960G
I'm shopping for good deals on Cisco phones. Forgive my ignorance, but I spent over an hour at Cisco's web site and Google trying to find a definitive statement as to the differences between a 7940 and 7940G phone. Anybody know? Should I prefer the "G" and why? Related question, is the 7960 worth so much more than the 7940? Has only 4 more buttons that I can see. Anything else under the hood that makes it worth that? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QOH (quiet on hold)?
Hi all, I'm not yet an * user, but I'm planning to become one soon. Here is a random question for you * experts: I'm often dialed into a teleconference using an external service (dial toll free number, enter PIN, etc). Sometimes I have an incoming call or want to call someone else and then return to the conference. I can't put the conference on hold because MOH will be injected to the conference (this is extremely rude when it happens because the entire conference is shut down until the holding party comes back!). So I'm stuck. What I want, in addition to MOH, is QOH (quiet on hold). Then I can put the conference on hold, no sound will disturb the other participants, and I can return later. Obviously I can make all holds be quiet (no music), but I would prefer to retain MOH as the basic hold function. The QOH would be an additional "feature". I'm likely to be using Cisco phones if that matters. So, can * do this, and if so, how? Can MOH be selectively enabled/disabled by extension? Are there other ways to solve this problem besides QOH? Thanks for everyone's help. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te410p with serial console fails with error:TE410P: Double/missed interrupt detected
On Mon, 25 Aug 2003, Gavin Hollinger wrote: > Thanks Mark, Sorry for all the questions, I am treading on > un-familiar ground here. Can anyone think of a way to adjust > how long the serial console holds the interrupt when the kernel > sends a message? Anyone know what the purpose of this behavior > is? Here are some uneducated thoughts: 1. Make serial port polled instead of interrupt driven? Majority of traffic is output that will be buffered. 2. Increase serial port buffer size? Perhaps problem is so much serial output that buffer fills up and blocks. 3. Make serial port faster? Would help drain buffer quicker. 115200 baud is routine, I've seen 460800 baud being used for embedded Linux on short cables. 4. Is there a way to have a remote console via ethernet? Won't help with BIOS settings, but would remove serial port issues. 5. Use different serial port hardware (USB serial dongle, PCI multiport card, etc) for console. Switch serial cable to do BIOS settings if need be. What BIOS do you have that is serial configurable? I'm always on the lookout for that. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 to T1 on asterisk?
On Mon, 25 Aug 2003, Mark Spencer wrote: > > Nothing like being on the cutting edge. :-) > > Always The pioneers have arrows in their *fronts*, the cowards have arrows in their backs! > > If that is true, then the bits are not mangled by * at all, > > answering the previous question. But ISDN is actually > > *easier* in some ways, no DSP on the samples to recover the > > modulation. > > The trouble with ISDN is specifically that there is no "echo > cancel disable" tone preceeding the call. How does the portmaster distinguish between an incoming ISDN call and incoming analog call? I know this can be done, my local ISP can handle ISDN and analog calls on the same phone number and it must know when the call comes in. Whatever method the portmaster uses to tell those apart should be applicable to * to disable the echo cancel. This might be as simple as "voice" versus "data" call (is that info provided by the PSTN?). Is the echo cancel needed on voice ISDN calls? I can live with no support for voice ISDN calls (can imagine why I would ever get one). -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 to T1 on asterisk?
On Mon, 25 Aug 2003, Mark Spencer wrote: > > 1. Will Asterisk route from one T1 to another "perfectly"? > > That is, the bits that arrive on the Portmaster would need to > > be the exact bits sent on the PSTN T1. Seem obvious that > > this should be so. > > As of this weekend it does. Nothing like being on the cutting edge. :-) What sort of change was required to make this work? > > 2. Would you predict any trouble interfacing a Portmaster to > > the Digium card? Can it both "sink" a T1 (from the PSTN) and > > "source" a T1 (to the Portmaster)? > > Yes. This should be fine. Might consider turning off echo > cancellation to be sure. In theory, the modems send the "don't echo cancel" tone to turn off echo cancellation in the connection, but I think you are suggesting that echo cancellation be manually hard coded off for that extension, right? In that case, is the incoming T1 digital stream identical to that handed to the outgoing T1? Should just be a copy inside *, right? > > 4. Will the above plan actually achieve the "56K" modem > > connection when routed through the asterisk box? That is, > > will there be issues of latency/bandwidth in handling the 64 > > kbps streams? > > In principle it should be able to pass even ISDN calls through. If that is true, then the bits are not mangled by * at all, answering the previous question. But ISDN is actually *easier* in some ways, no DSP on the samples to recover the modulation. Thanks all for the help! -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 to T1 on asterisk?
Hi all, To solve my need for dial in modems, I've hit upon an idea: buy a used T1 "analog" modem bank like a Lucent Portmaster that takes in a T1 and provides several 56K modems. This is overkill for a lightly used dial in service, but the prices of these boxes is so cheap (~$300) with ISPs going away from dial in service that it makes sense. This is how I imagine it would work: Asterisk PC with one TE410P 4 port T1 card. One port goes to the PSTN and has my incoming calls. One port goes to the Portmaster which has perhaps as many as 24 "modems" in it. Two ports are free. A user would dial in, get the auto attendant, dial in an extension (for the modem), and be transfered to the Portmaster T1. The Portmaster would establish a 56K modem connection with the caller. Some questions for the gurus out there: 1. Will Asterisk route from one T1 to another "perfectly"? That is, the bits that arrive on the Portmaster would need to be the exact bits sent on the PSTN T1. Seem obvious that this should be so. 2. Would you predict any trouble interfacing a Portmaster to the Digium card? Can it both "sink" a T1 (from the PSTN) and "source" a T1 (to the Portmaster)? 3. Has anyone out there done this? Any suggestions? Which modem banks to buy/avoid? Places to buy used/surplus/cheap ones? 4. Will the above plan actually achieve the "56K" modem connection when routed through the asterisk box? That is, will there be issues of latency/bandwidth in handling the 64 kbps streams? Thanks for everyone's help. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial in modem speeds over VoIP?
Hi all, I like to have a dial in modem on my toll free number so that when I or my employees travel, they can always get in for net access to read email if no better method is available. Right now, my Panasonic KX-TD1232D PBX receives the call on a POTS line and routes it to an analog modem. The speeds achieved are around 24 kbps due to the digitization of the KX-TD1232D introducing quantization errors (it is a "digital" phone system, which means it digitizes on the way in and the way out). Now I am considering moving my toll free number to a VoIP service like nufone. But how does that affect analog modem calls? Consider an analog modem somewhere in the PSTN calls through a toll free number handled by a VoIP service that is then routed to an * box at my place of business. The * box routes the call out an FXS card to an analog modem. What sort of modem performance should I expect? The signal gets digitized by the local CO at the calling site then sent to the VoIP provider (presumably as 64 kbps ulaw encoding). They potentially compress the signal using some codec (which may be quite a bit less than 64 kbps in data rate), potentially introducing delay jitter, and certainly introducing latency. Then I get the voice in packets and send them out an FXS port on the * box. This introduces additional delay, jitter, and noise. Sounds like the modem might not even be able to train much less transfer bits. Does anyone have experience trying this or educated insights into how well this would work? I imagine that many residential VoIP providers (say Vonage on ATA-186 boxes) must have had modems connected to them. This would be similar effects but in reverse. Does Vonage have guidelines/comments about using modems on their service? Has anyone tried this? Related question: is there any way to avoid going out the * box on FXS to an analog modem? That is, do we have software that can DSP the audio stream and recover the modem data bits directly? A kind of virtual modem module? That might make it much better as it avoids one more digital to analog to digital conversion. It would essentially be the code found is so called "soft modems" but taking it's input from packets rather than sampling a phone line. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Provisioning CO lines
On Thu, 21 Aug 2003, David Carr wrote: > I discovered and deployed a solution some would consider > counter-intuitive. I love "out of the box" thinking. What kind of business is it? > For whatever reason, I can get a dedicated long-distance T1 for > about $400 MRC ($16 per line) >From who? > For the dedicated long-distance T1, all inbound and outbound > calls cost 1.9 cents per minute, even if you are calling a > toll-free number or the company next door. Instead of having > local DIDs, we simply got a huge supply of toll-free numbers, > all with unique DNIS so we use them as DIDs for not only direct > voice numbers but also direct fax numbers. For businesses with national or global footprint, this makes sense. You're not bothered by dialing all the digits to go out (since local calls are rare) and your customers are not bothered to dial in on toll free numbers (since they aren't local either). This simply means *all* calls are $1.14/hour which isn't much. > Do the math for your company. To get the T1 and LD rate, how many minutes/month does your business do? Clearly, you are getting some sort of volume discount that smaller business might not be able to match. I wonder: 1. Does this violate some FCC rule since you're not paying certain fees (which are tied to local lines) on your LD T1? 2. How does this handle 911? Perhaps a POTS line and route "911" to it when someone dials it? This appeals to me given the cost and legal burdens placed on local lines. But it won't work for everyone by any means. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Provisioning CO lines
Hi all, This is a NEWBIE question, so all you experienced types that are tired of stupid questions can move on... I've pretty much given up trying to do my entire phone system over IP (including local service), so I have to select and provision my local CO lines. I need about 10-12 lines which can be POTS lines, of course. But, I thought, why not get something digital and expandable like a DS1, PRI, T1 or whatever they call it with 23 or 24 channels of 64 kbps voice. It seems like it would be simpler for me to deal with this (and better quality) and it *should* be simpler for the phone company, too. However, while everyone can sell me POTS lines, when I ask about getting these in some sort of digital muxed interface, I seem to confuse the providers. In one case, I was able to get something called "channelized T1" which cost a lot and did not actually include the "phone" service for any of the channels, that was additional. So the cost to go from POTS lines to something digital was extreme, so much more than I can't understand why anyone would have T1 voice interfaces, yet all the PBXes have this and it seems commonly used. I must be doing this "wrong". Okay, so I need help with: 1. Understanding terminology so I can ask for the "right thing". 2. Advice on when it is reasonable to go POTS versus something else and what that something else is. 3. Feedback on what others are doing with 10-12 lines in the US that may want to expand to ~20 lines. 4. Interfacing so many POTS lines to Asterisk. I guess that means an FXO channel bank to T1 card? Kind of stupid to go digital/analog/digital in the last 100 feet. Help? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP dialtone?
On Wed, 20 Aug 2003, Jeremy McNamara wrote: > NuFone doesn't restrict any number of simultaneous channels and > we do have a wholesale platform we ~can~ offer. How do I find out more about this? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP dialtone?
On Wed, 20 Aug 2003, Ernest W. Lessenger wrote: > At 04:48 PM 8/20/2003 -0500, you wrote: > > >Now, if that is possible, how does the VoIP dial tone provider > >get my inbound local and toll calls? I would want my "local" > >phone number to work, of course. > > You would need to redirect your local number to them. This > ALWAYS assumes that the VoIP provider has a switch in your > local CO or an agreement with someone who does. Vonage and > Voicepulse, for example, do not have a presence in my area. I > intend to maintain several POTS lines for incoming calls, and > use a VoIP provider for all outgoing calls. Oh well. I'm would expect no one would have presence here. This sounds so suboptimal, you have to provision *two* systems, one for inbound (local CO) and one for outbound (VoIP provider). Of course, the outbound can be just your internet connection, but this still seems annoying because most of the money is in the local CO service. Hmm, perhaps *all* incoming calls can be toll free? I would maintain the one local CO POTS line for 911 out bound, and then only use my toll free number for inbound. For the money I would save on local CO lines I can buy a *lot* of toll free minutes! Then the VoIP dial tone provider can route my toll free number to me over the internet. Presumably, then, there is no real limit on the number of "lines" coming in. It isn't hard coded like the CO lines are. This all seems pretty fanciful at the moment... -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP dialtone?
On Wed, 20 Aug 2003, Adam Roach wrote: > > I guess my question was a little deeper than that. > > Can I simply ditch the PTSN? > > 911 is the sticking point. Ah. > Until this tiny, possibly life-or-death detail gets sorted out, > I'm probably going to have at least one traditional phone line > at all times. Hmm, okay, so would it be possible to maintain *one* POTS line that is used if anyone dials "911" on their desk phone (set this up in * dial plan), then it connects to emergency services properly, and then use a VoIP dial tone provider for *everything* else? This assumes we are having only one emergency at a time! Now, if that is possible, how does the VoIP dial tone provider get my inbound local and toll calls? I would want my "local" phone number to work, of course. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP dialtone?
On Wed, 20 Aug 2003, Brian West wrote: > I think NuFone can do what you need contact [EMAIL PROTECTED] > > I have inbound 800 service and outbound ld service with them.. > works great. And for local service, you do what? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP dialtone?
On Wed, 20 Aug 2003, John Todd wrote: > At 3:20 PM -0500 8/20/03, Mike Ciholas wrote: > > >Are there VoIP dialtone providers? That is, could I use only > >my internet connection for voice calls and not have a separate > >T1/POTS bank for that? > > >First question: Does such a thing exist? Where? > > Yes. > > >Second question: Does it work? How well? > > Works great. I haven't made a long distance call on my PSTN > line in months, and I spend pretty much all day on LD calls. I guess my question was a little deeper than that. Can I simply ditch the PTSN? I see that toll free inbound and LD outbound can be handled, can they handle inbound and local, too? Seems like we are very close to cutting the local phone company out of the loop! That would be so nice as trying to talk them about provisioning the lines is quite a chore. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP dialtone?
Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone. Calls come in and get VoIP'ed and sent to me as packets, perhaps IAX, perhaps something else? First question: Does such a thing exist? Where? Second question: Does it work? How well? Third question: Would you want it? Why? Fourth question: How much $$$? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk ready for "real" use?
On Wed, 20 Aug 2003, Scott Lambert wrote: > RJ11 plugs work in RJ45 jacks most of the time. Yuck! I was told, and I have verified, that when RJ11 goes into RJ45, it overbends pin 1 and 8 contact wires so they don't "relax" back to their original form. This happens because there is no "slot" in an RJ11 connector for those contacts, so the plastic bends those wires much more than the others. This causes loss of contact pressure when a real RJ45 plug is inserted. You can tell when this happens by looking at an empty RJ45 socket. If pins 1 and 8 don't line up with the others, someone has plugged in an RJ11 into it. Sounds like a way to have flakey RJ45 jacks all over the place, and ethernet does use pin 1! This must be a FAQ somewhere... -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk ready for "real" use?
On Wed, 20 Aug 2003, Ernest W. Lessenger wrote: > At 10:42 AM 8/20/2003 -0500, you wrote: > >So, I would like to hear from those out there that have a > >system as I've described above and tell me if I'm insane to > >commit this direction or whether it makes sense. > > I'm in almost the same situation as you. However, I'm mostly > worried that the customer service desk here will start to > complain that they can't tell how many calls are in the queue > any more (our current phone tells us how many calls are > ringing, on hold, etc). Regardless, I'm very interested to hear > your results as well as what others on the list say, and would > like to stay in touch with you if you decide to move forward > with Asterisk. Needs like yours are probably why you *should* choose Asterisk. Sounds like "there outta be a way" to do that. Right off, I can see keeping queue length counters and displaying them on the phone, or perhaps having them displayed on a dynamic web page visible to all. You could imagine doing things as sophisticated as "your expected wait time is X", when the queue grows over X size, call other extensions, etc. I am very intrigued by the flexibility Asterisk offers, but I need to know that I can reliably just "make calls" at first. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is Asterisk ready for "real" use?
Okay, I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our "traditional" PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. It is *not* an option to purchase a VoIP system package from Cisco, 3com, etc. Installers are getting an enormous premium for this now (rough estimate, 20 extensions $40K (!)). I am "this close" to committing to a solution based on Asterisk PBX, PoE LAN switches, and VoIP phones. I am absolutely sure it is the right *long term* solution, but I don't know if it is "ready" for reliable daily usage. I've literally read the last year's worth of posts to asterisk-users to get a "feel" for the situation. Since you don't see posts of the form "installed it, just working, no problems" very often, you could get the opinion that everyone has problems since that is what the mailing list is for. So, I would like to hear from those out there that have a system as I've described above and tell me if I'm insane to commit this direction or whether it makes sense. For those of you who have done it, how much time did it take you to get the system running smoothly? PS: In case it matters, we're extremely Linux capable (we use it for our file serving, networking, and we built our own custom ERP on perl and mySQL, we also do embedded Linux in custom military robot controllers). -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: LAN switches with PoE? PoE phones?
> From: Steven Critchfield <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Date: Mon, 18 Aug 2003 12:45:19 -0500 > Reply-To: [EMAIL PROTECTED] > > It is possible to make your own PoE adapter built into a punch > down block. This is if you can find an appropriate 48volt power > supply. I built an adapter for the 7960 we had here in the > office, but my boss preferred the wall wart idea. Hmmm, I bet that violates the PoE specs as there is nothing "controlling" the 48 volts. That is, it isn't looking for a valid PoE signature in the terminal device before it sends 48 volts down the wire. This will surely fry some non PoE ethernet devices. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LAN switches with PoE? PoE phones?
Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of "power hubs", but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be special for IP phones in anyway? QoS support? Managed? Also, are there PoE phones that work with *? Most I look at seem to be powered from AC wall blocks. We'd like to centralize the switching and power and provide a UPS so the phone system works when the power goes out. [Apologies, I'm new to this whole concept of IP phones and *.] -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
