Re: [asterisk-users] Call parking with ISDN
Since no one has responded to this, I am wondering if there are two kinds of call park. I haven't worked with European ISDN, but if it has a call park feature, that would be distinctly different from the Asterisk PABX call park feature. The Asterisk feature should not matter what sort of trunk was involved, which is why I am wondering. On the other hand, if there is an ISDN park, I'm not sure Asterisk would support it. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MISDN/asterisk problem (not sure where from)
Great. That is a configuration I have not heard of before, but again different countries do things differently. Good luck with the ISDN part. I haven't gotten ISDN and Asterisk working here, yet, primarily because ISDN is configured differently in the US and not widely used, so if a solution exists, it will take more digging than I have been able to do so far, and I haven't found anyone who can even point me in the right direction to start. You should be better off, as the European version is much more widely used. I definitely consider it a better interface to a digital system than POTS. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MISDN/asterisk problem (not sure where from)
The extra information does help. Unfortunately the Samsung 3010 is not available here and the information I could find about it was all in German, which I don't know fluently enough to read. I also run the risk of assuming standards that are in fact different in different countries. The only DSL installations I am familiar with do indeed deliver telephone service as an ordinary analog line (POTS). The internet service is on a carrier so that all energy is above the audible range. The phone can plug into the router so that a filter can split the signals apart. That way the phone won't "short out" the higher frequency DSL information. At least in our system, it is also possible to place filters at other locations, such as in phone jacks to avoid having to route everything through one location. The part about the ISDN s0-bus is not familiar to me. It sounds like it is for configuring the router. I doubt that it has anything to do with making voice phone calls. The basic architecture of DSL is to superimpose a data carrier on top of an analog voice line. Again, a simple test is to connect an analog phone to the line coming from the phone company before it gets to the router. If it works, the voice service is analog--plain and simple. The test will probably temporarily disrupt the internet connection (in fact the simplest way is probably to unhook the router and replace it with a phone). If this indeed proves to be the case, you need an FXO of some sort, not an ISDN adapter and Dhadi, not mISDN. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MISDN/asterisk problem (not sure where from)
I may not be the best one to answer this because it sound like European ISDN, but there seem to be some basic issues that might be relevant >a Samsung router with analog and ISDN ports. I'm not sure what this has to do with the voice side of things. An ISDN port on a router would generally imply that the purpose of the router was to pass internet data over an ISDN, something that was quite common a decade ago, but has largely been supplanted by higher speed methods (DSL, Cable modem, etc). Some such devices did have the ability to pass voice traffic through to an analog port (FXS style). I use such here in my home office, although at the moment, strictly for voice. >the phone company says the outgoing line is analog landline That is easy to test, hook a phone directly to it and see if you get dial tone. If so, then there is no place for an ISDN router in this except if you are using the analog line as a dialup backup for data (assuming the router supports that). >but I'm sure it's some VOIP. Oops, we have a third technology that popped up here from nowhere. VOIP is carried over the internet or a LAN/WAN. It has nothing to do with either analog voice or ISDN. >From here the post heads rapidly downhill and out of my area of expertise. If any of my above statements are true, you are totally barking up the wrong tree and would be equally well served by trying to attach the phone line to the antenna connector of your TV (don't try it, it wouldn't be good for the TV), hence any error messages, etc. have no bearing on the problem. If you indeed are getting ISDN from the phone company, then reposting a more specific scenario may get you help from someone more familiar with your countries protocol. In summary, you must start by determining if the phone line is analog (easy to test), ISDN, or some sort of broadband data which would support VOIP. Once you know that you can connect it to the proper interface card. At that point if it doesn't work right, someone here may be able to help you with a configuration file or something. At the moment what I have stated is about all anyone can tell you. You appear to be quite a ways away from knowing what you have. Wilton Helm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk work here
The only thing I know about the T1 is that it uses wink start signaling. Wink Start? That is an analog protocol used by DID or E&M trunks. If that is what it is using, then the T1 must be a digitized set of DID analog trunks. A wink is a hook-switch-flash used to tell the originating side that it is ready to receive DTMF digits. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
[snip] >You may be able to boost the battery voltage with a simple dc adapter in >series to get the line build out capability you need. Just make sure its >floating with respect to ground and wire it in. Don't be afraid of >hurting the phone, you won't. But it is possible to hurt the ATA if it causes more current than expected to flow. It is even possible to force reverse voltage into the ATA that way. A line fault would make it even more likely. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
You are exactly right. Cat 5 had no advantage over cheaper wire for voice, and the length limitations are meaningless. Consider that Cat 5 is typically use with signals that extent to 30 MHz or beyond. A voice grade analog circuit must go to 4 KHz (1/10,000 as much). At 4 KHz, the wire generally doesn't even act like a controlled impedance. Wilton From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 8:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port I could be wrong but I don't think the cat5 limit of 100 meters applies to any analog signaling over that copper. I believe it only applies to Ethernet signaling. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
There are a lot of factors that impact this. First, CAT 5, while usable is overkill. Cat 3 (otherwise known as I/O wire) works equally well for voice grade lines. That being said, for that long a run, a heavier gauge wire would be better. I believe telcos use 18 - 22 guage (Cat 5 and Cat 3 are both 26 awg). This has less resistive loss. Most FXS or ATA devices use 24 volts or less for "battery". That works fine for short loops, but limits the range. A central office POTS port normally uses 48 VDC which works well to several KM. If the customer is at the end of a long run in a rural area, they use a "long line" card which uses 75 volts. (In rural communities, they often place the line cards in a roadside "remote terminal" and use statistically multiplexed T1s to make it appear to the switch as a part of it. That addresses the DC characteristics, which can be reduced to ohms law. A phone needs around 8 V @ .02 A. The wire resistance determine the drop (E = IR) and the source voltage determines whether there will be enough left. The A.C. characteristics are more complicated. The FXS must do a 2 wire to 4 wire conversion, which involves matching the impedance of the line. The FXS is generally designed for relatively short lines, so might not be able to match either the resistance or capacitance found in a long run. Heavier wire will minimize this. In addition to that, the transmit side of the 2 wire to 4 wire circuit must be able to drive the load it sees, and again it may not be designed with a long run in mind. Finally, COs line cards have the ability to adjust receive and transmit gain to compensate for sound level losses in long lines. While this isn't routinely done on simple circuits, it is an option an FXS doesn't generally have. In addition, the more gain that is inserted, the harder it is to balance to 2 wire to 4 wire circuit, and the more complex it has to be in order to support this. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
>BTW, can someone explain to a libart major like me (;-)) where echo comes on in a telephone conversation? I seem to recall it's due to the length of the line between the CO and the local party, but I'm not sure. Yes, I'll tackle that. It takes a finite amount of time for the electrical signal originating in one phone to arrive at another phone over whatever path it is taking. If the path is copper, that time can be fairly small. If the path is satellite, the time will have to exceed the 1/4 second round trip to the bird. If there are SIP packets involved, the time must be larger than twice the packet size because of the time taken to collect the data in the packet and then to serialize it at the other and after it arrives. If the path involves the internet, there is the path delay there to be added in (ping will give you an idea of what that is, but it can often be 50 - 200 ms). All of this constitutes a delay. It can be a bit annoying in its own right because one person asks a question, and twice the delay time elapses before they start hearing the answer. However, if there are POTS analog circuits involved anywhere, a second factor comes into play. A POTS analog circuit is two wires, which carry an electrical representation of sound. Both sides of the conversation are carried over the same wire. (its called a 2 wire circuit. There are also four wire circuits where each direction travels on a separate pair of wires. They don't have echo problems. Digital circuits also have separate paths for each direction, so are immune to echo) The problem with a two wire circuit is how to separate the sound going in both directions. That is done by something called a 2 wire to 4 wire converter, also commonly known as a hybrid. It basically works by subtracting out what it knows is being sent at the near end from what it sees on the wire. If that subtraction is perfect, only what came from the other end is left and that is presented to the listener. In the real world, this isn't perfectly possible, but it can be done fairly well. However, there is a side effect that comes with the transition from two wire to four wire. Some of the signal originating at one end of the wire gets to the other end and is reflected back. For an analogy, tie the end of a long rope to a pipe, stretch it out and snap the other end. You will see a wave travel to the pipe and then come back. If you were able to attach the rope to the pipe with a suitable dashpot or something that would fully absorb the wave, nothing would come back. This reflection from the other end is the cause of echo. If the path is terminated in exactly the correct impedance, there would be no echo. However, for real circuits over the range of frequencies that make up sound, that impedance is a complex quantity, and cannot be exactly matched. The bottom line is that any circuit with one or more 2 wire analog portions is going to have some echo. Since most of the circuits provided by a phone company are POTS, they are two wire analog from the subscriber to the CO. If the subscriber equipment is Asterisk, then a 2 wire to 4 wire conversion and digitization takes place there. Likewise virtually all telco links are digital and a conversion takes place in the switch in the CO. Then at the other end the process is repeated. That makes a total of 4 interfaces where echo can originate in a typical phone call. If part of the call is SIP, or internet or satellite, the delay is large enough to guarantee it will be noticeable. Since there are several interfaces there can be several echoes. Another example that illustrates the concept is a speaker phone. If the person on the other end is using a speakerphone, then some of what you say comes out of the speakerphone, bounces off the walls of the room, gets picked up in the mic and comes back to you. Again, if the delay is very large, it will be an echo by the time it gets back to you. Speakerphones (if they are full duplex--i.e. allow both parties to talk at once) have to have echo cancellers to prevent this from happening. >Is there a way to keep track of this issue, and overtime, to configure it to answer a call by expecting such and such echo, and thus, avoid starting sampling from scratch every time? Unfortunately not. If you've followed the discussion to this point, you understand that the magnitude (loudness) of the echo depends on the impedance mismatch which is unique to the circuitry at each end (for a typical call) of the call. The delay time is unique to the call path, which is likely different for each call, and in the case of internet calls, can vary within the call. The echo canceller must constantly do pattern matching to recognize changes and adjust for them. Its job is to subtracting out a signal of exactly the same amplitude as the echo, but of the opposite polarity and delayed by exactly the path delay the echo is travelling through. Since there can easily be four or more ec
Re: [asterisk-users] cheap CHEAP ata
>Have you checked ebay? Just beware that there are a lot of ATAs on Ebay that are locked to Vonage or similar providers. While they are not impossible to unlock, it requires considerable time and good Linux networking experience, as the process generally involves creating an isolated world (with its own DNS, etc) that mimics the provider and then "updates" configuration files. If you want a lot of cheap ATAs it might be worth your while to set up such a system, as most of the work would be for the first one and the rest would be relatively easy. On the other hand, if you weren't anticipating this problem, you might get stuck with a bunch of useless paperweights, which wouldn't make the total cost of the solution very cheap. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Practice Advice?
>Also, FC10 is out. You should probably grab that first. Unless you are a strong Linux Guru, I would never recommend a Fedora release for a production system. I have FC9 here and FC10. It took me months to eliminate the bugs from FC9, and I still haven't gotten FC10 to install on the machine I got it for (three months now). Fedora is cutting edge and puts out a new release probably every six months with less than usual regard for consistency or stability. I don't know of anything Asterisk that requires this level of cutting edge technology. While all the bugs I fought in FC9 are gone, they have been replaced by a whole new spate of (some still unidentified) bugs. Centos is a much more appropriate distro for production work. Nothing goes into it until it is known to be rock solid, and update occur much more slowly. It wouldn't be too far off base to say that Fedora users are the beta testers for Centos--not explicitly in terms of versions, but certainly in terms of features and code base. I'm sure there are other good (maybe even better) distros for Asterisk, I'm not familiar with all of them. Fedora is really at home with someone who is running a personal web server or media computer or something for a hobby and likes to have the latest of everything and wants to (or at least is willing to) play with it, get it to work and help improve it. That isn't the recipe for running a business. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems?
>If half-duplex audio is good enough for you, sure. You've lost me there. I am not aware of a modem that is for sale today that is half duplex. (OK some support a couple of minor half duplex modes). All state of the art modem protocols send and receive simultaneously using the full 300 - 3000 Hz bandwidth in both directions with adaptive equalization and echo cancellation to make it work, which is pretty much what a voice circuit need. There are two differences: 1) The response and quality of a current modem must be considerably higher than what is needed for voice use or it would never achieve the throughput expected of it, and 2) the adaptive equalization algorithm is designed around modem specific techniques. The latter is (especially for a softmodem) a software issue, not a hardware limitation. >Only a fraction of the hardware available is actually capable of full duplex >audio. Absolutely not the case. Particularly the softmodems (the most inexpensive) contain little else than what is required for placing and answering full duplex audio calls. Everything else is in the driver. The OP is 100% correct, that they would be an excellent candidate for FXO use in low volume applications. >What it really comes down to is a value proposition: Quite true. This is the real issue. As mentioned, these drivers require considerable skill and knowledge to write. While there is no doubt that the result would be very cost effective, the business model is lacking. The modem manufacturer is going to see the potential market for this as somewhere down in the noise compared to their normal modem sales, so isn't inclined to invest. A third party developer with the skills would have a difficult time recouping development costs (let alone any profit) because they don't control the hardware, and therefore have no leverage. A user with enough volume to justify paying for the development (or doing it if they had the skill) probably has enough volume to use T1s instead. If everyone that could benefit from using a modem card were to pitch in $10 towards the development, it would probably be quite possible. But how to make that happen? Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T-na Bricked?
If there is such a thing as a complete Reset, the mfg. is mum on it. The NA is supposed to be the generic open version. However if a provider chose to lock it up, they can make it extremely hard to get into. Vonage routinely does this to their PAP2s, (not NA). You can Google the topic and get some insights. Nothing is crack proof, but if someone has done something similar to what Vonage does, you have to get a bunch of information about the unit and create an isolated LAN that looks like the providers host and convince the device to get updated information from it. I've read most of the material, but decided it wasn't worth my bother to jump through the hoops on mine. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hum noise
Yes, you make a good point. Electromagnetic fields are another source of ingress, whether from a nearby cell phone or by being located a mile away from a 50 KW AM radio transmitter (etc.). >one does wonder why there's such inadequate shielding As a ham radio operator, I can say that has been a question raised about consumer electronics in general for many decades, particularly since the advent of solid state devices with their low voltage diode detecting capability. The answer, unfortunately, is simple--economics. If only 1 out of 1000 devices is exposed to a situation that results in interference, the manufacture is loathe to spend even a nickel to protect against it--especially if the volume is large. Five cents on 1,000,000 products is $50,000. Depending on the strength of the interfering signal, it can take a lot more than five cents to protect it. Cell phones aren't very high power, The 50 KW transmitter--well that's another story. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hum noise
There are a large number of potential sources of hum and each situation will narrow them. The first thing would be to quantify the observation. I am assuming it is power line frequency, although that may not be the case. It is also useful to notice whether it is fairly pure or rich in harmonics, or even if it is predominately the second harmonic, which would be produced by full wave rectification in a power supply. Classically in telephony, hum is caused by imbalance or things like split pairs that disturb the balancing capability of the twisted pair. However, as has been pointed out, hum can only enter a system at an analog point, and the OP indicates that the only analog point in the system would be inside the SIP phone, unless there is something analog beyond the T1, but that is probably outside of his control. Since Asterisk is connecting a T1 to a SIP phone, there is no way anything inside the box (PC) can be the cause of the hum. If it had FXO or FXS cards in it, magnetic coupling or even electrostatic coupling inside the PC would definitely be a consideration, and would most likely manifest itself as some sort of buzz that was NOT related to the power line frequency. The comment about the wall transformer on the SIP phone is very germane. If it has a bad or inadequate filter capacitor that could allow hum into the SIP phone through the power voltage. It would most likely be 120 HZ (US, 100 Hz in many other countries) and would have additional harmonics, because it isn't sinusoidal. One test would be to power the phone from a known good laboratory type power supply as a test. The other main culprit in this case would be coupling of magnetic fields into the phone itself, either the electronics inside the housing, or the handset components, or even the handset cord. Proximity to any electrically powered device with a large transformer would be a potential source. Even a nearby CRT terminal or monitor. An easy test for this would be to move the phone and/or handset. Generally just rotating it 90 degrees will make the loudness of the hum change noticeably. If that is the case, then identify the offending source and move or replace it. This situation is actually simplified because the SIP phone is the only thing that could produce the symptom. Its just a matter of determining how it is entering the phone, through the power supply or directly into the electronics from a nearby source. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a public blacklist of hackers'IPaddresses?
Interesting thread. I am not doing this commercially, so I don't know all of the issues at stake. My initial reaction was, "what problem"? But, subsequent posts have clarified that some. I do see some mitigating factors though, particularly re the banking model. First, telecom providers aren't generally dealing with large amounts of material susceptible to identity theft the way many other businesses are, nor are hackers generally looking there for such. The main potential loss I am aware of, and that has been discussed here is provided services. The impact of that depends on the model a particular company is working on. The worst case is a re-seller who has to explicitly pay for each minute used/billed. Other providers are paying for bandwidth, but that is more nebulous. Sure, a provider makes money by selling minutes. But the guy in China that hacked his way in isn't going to buy minutes of his hacking is denied, so there is no loss of potential revenue, only loss of available bandwidth. If that bandwidth is significant it should raise an alarm, which one would hope would cast light on the "leak" and cause it to be discovered, rather than the available bandwidth increased. If the loss is not significant enough to draw attention to itself it may well be a minor cost of doing business. The OP mentioned insurance. I'm not sure, at least in many cases, if the amount of potential hard cash liability exposure is sufficient to warrant insuring. If someone is getting hacked to the tune of 10% of their bandwidth or revenue, and doesn't have any way of noticing the problem, they probably aren't qualified to be running such an operation. One relevant example from the banking industry. About once a year I get a call from one of my credit card providers wanting to know if I indeed made such and such a purchase at such an such a location. Their potential exposure is very large and they do continuous, fine tuned profiling. They know I don't live in Australia and if they start getting charged from companies in Australia, they want to know why! They have it a bit easier, because they have more information to work with, but there are certainly things that can be profiled. Most users are going to originate from one or a small number of IPs. Some may originate from every Starbucks in the state, but that's a recognizable pattern. Fortunately most hackers don't know that profile and won't necessarily steal the account information of someone who has a profile like they do. Also, they tend to "call their girlfriend in Mexico 50 times in two weeks", which is hugely different that the real user does. If nothing else, identity thieves (this is a form of identity theft) tend to use the stolen identity as much as possible before it gets discovered and stopped. That alone is a major profile difference from a typical user. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?
If life were only that simple. A lot of hacking passes through unsuspecting intermediary computers, precisely to hide their tracks, not to mention IP spoofing. People have offered for sale access to 10,000 computers to use for propagating mischief. That's a lot of IPs to block! I got hacked about six months ago. They came in through SSH and figured out roots password, which was a concatenation of two English words. I presume they did a dictionary search. Then they changed the password, replaced some key files and launched a denial of service attack against somebody (including compiling the program on my machine)! I traced the IP address to a Comcast customer in Indiana or something and notified Comcast, but haven't heard anything. Probably their customer never even knew it happened--it was probably a hijacked situation. Prior to that I had been logging hundreds of robotic attacks a day that were unsuccessful! I re-installed everything and changed my SSH to a non-standard port and used a more robust password. I haven't had a single hack attempt the four months since. For my purposes, I don't really need SSH on a standard port. That made all the difference in the world. Two areas that have had large hacker presences in the past: Russia and China. A lot of E-Mail spam originates in those two areas, also. I've considered blocking the entire host domain for any provider generating spam from those regions, as I have no legitimate business need to correspond with people in those regions in general. However, I suspect it might block messages from a few users on this list, and I know it would block at least one user from another list I am on. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
>12kHz isn't really enough for high quality voice, and the extra bit >rate needed to push the bandwidth to 15kHz is small. Also, a deep man's >voice looses something when you cut off at 70Hz. I'm not sure that this isn't stretching things a bit. There are no handsets or headsets (AFAIK) that can do justice to 50 KHz and probably most speakers attached to a PC can't. Likewise, while a deep male voice can go below 70 Hz, few transducers can do justice to those frequencies, either. I don't think the attempt is to reproduce a symphony. The extra bandwidth (even if it is minor) would be hard to justify if one needed $500 speakers to benefit from it. While a number of people might be able to tell the difference in an A B comparison, I suspect few would notice it without direct comparison. I also suspect Skype is correct in that the majority of people, listening to it on typical hardware would like additional low frequencies less than without because of things like distortion in the transducer. Getting the bandwidth above 3 KHz at the top will improve intelligibility, but somewhere between 5 and 10 KHz that reaches a point of diminishing returns. Likewise, extending the low end below 300 Hz will help naturalness, but that also reaches diminishing returns somewhere around 100 Hz unless all the pieces are very high quality (from the mic to the speaker). It seems to me that they have exceeded those realities by a comfortable margin, which is generally what good engineering is all about. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building a phone
>This is not entirely true - many of the nokia phones use a java OS as a >core, and you can load pretty much any java software you want on them, >but all the points about power and battery use are still valid. (and >whether you really consider that truly an OS is questionable, but its >out there) Java is the worst offender. Its resource requirements often exceed those of the application it is running. Java is useful for things like displaying web pages that are not time critical and where its write once, run everywhere philosophy is valuable. But anyone trying to actually do things like I/O control, call setup, transcoding, etc. in Java are asking for every issue I raised. If WCE can get 8 hours of battery life, Java would be about 3. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building a phone
>Again, the main reason for me to require a higher end CPU is audio >compression. But I also want the system to be run by a standard OS. It >needs to be easy to add your own application there. Mutually exclusive. I don't know any standard OS that doesn't waste about 10 x as many CPU cycles as a SIP phone should ever need. The problem is generalization. A standard OS is designed to support a wide variety of devices, including a wide range of screen sizes. The abstraction layers that make this possible often consume more CPU resources than the application they are supporting. Most of that isn't needed for this application. Compatibility with WXVGA isn't required. Even a full blown file system is a luxury. Linux is about the closest thing because it can be pared down. But it takes someone with considerable experience to know how and what to trim. I supervised a system that used "busy box" to create a compact system that lived on a small flash card an some RAM. As an example, I have been a Palm owner for a number of years. I laughed when the Win CE stuff came out to compete. The Palm OS was written for the task at hand. I could go a week or more on a charge. The Win CE devices had to be recharged after 8 hours! Why? The OS required too much which required far more compute power, which ate batteries. The SIP phone you propose could be done with about 1 W of power plus a couple more for backlighting. An OS based version would start at 5 W + backlight and could easily go to 15 W or higher. Not the end of the world, I suppose on a desk (if there aren't a hundred of them and I'm not paying the electric bill) but a huge difference if it has to run on batteries. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building a phone
> I assume that the relevant application requires some non-trivial CPU power. I > would > exclude e.g. a 486-based systems. I'm not sure that's the case. The industry has gone in the direction of throwing lots of silicon at a problem, often as an excuse for poorly written code, sometimes in an interpreted language. There are a number of high integration CPUs out there that I suspect could do this sort of thing. I develop device controllers for a variety of industry needs. They tend to have Ethernet, RS-232, sometimes 1 Mb/s synchronous communication. G711, quarter VGA color LCD with touchscreen and control loops running at about a 1 ms rate. The entire code takes less than 256K in C. My choice of processor is the DStni Ex (made by Lantronix and sold by Grid Connect) which is a high integration, high speed 186 core with two 10/100 Ethernet Ports and 256K of RAM on it in addition to the usual assortment of other stuff. The above required platform adds three support chips (one being the LCD controller). The CPU can run over 100 MHz. Memory accesses take one clock and typical instructions take two or three. Cost is in the $10 to $20 range for the chip and power consumption is around 1 W (the LCD backlight takes more than that!) I'm sure there are several other comparable platforms out there, such as by Digi International. The Geode is a good candidate as are some VIA chips, if one wants to use protected mode x86. The biggest thing for this is don't even consider Intel. For most of their life they have not provided cutting edge solutions for embedded use. Most of their stuff consumes too much power. And most importantly, they are targeting the very volatile, short lived PC market. By the time you get an embedded design up and running and reach market penetration, you won't be able to buy the chip any more. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA recommendation (wih FTP provisioning)
>I don't think they are "locking" the same device that you buy when you buy >the "-NA" version. I believe that Linksys is making pre-configured >devices for these large buyers and selling them much cheaper to them in >bulk than they sell the -NA version to the community at large. I'm sure you are correct. The basic firmware is the same, but they come pre-loaded with configuration that automatically goes to the provider's web site. Again, that's not bad as long as there is a "factory reset" that can restore it, which there is, but it is password protected and the provider won't give out the password. If the device were owned by the provider, that would be acceptable, but the provider sells them to the customer so the customer deserves to be able to use them for other purposes if they need to. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA recommendation (wih FTP provisioning)
>Actually you want to use the one with -N/A at the end. Thanks for the correction. My memory failed me. But definitely I want people to know that not every PAP2T is useful to them so they don't get burned. IMO Vonage (and others) should not be locking them the way they do (and Linksys shouldn't make it possible). I don't mind them providing autoconfiguration, as many of their customers would be lost otherwise, but blocking factory reset and other tricks they play is IMO morally wrong, particularly since the customer paid for the ATA. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA recommendation (wih FTP provisioning)
Just beware that most PAP2s on places like E-Bay are from Vonage. They lock the things up quite seriously. There are procedures out there to unlock them but it requires stuff like setting up an isolated LAN with a DNS server and FTP server and a special file. If someone would make a live boot Linux disk that had it all set up it might be a useful service, but in the mean time, it wasn't worth my time, so I just wasted some money. The one with the letter T on the end is supposed to be general purpose and open. A lot of people don't know the difference until after they've wasted their money. Also they make the mistake of putting the thing on line immediately and it goes to Vonage and get the latest upgrade, making it even harder to unlock. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID's in a specific rate center
>> What is illegal is to set caller-id to a fraudulent value such that the >> person on the other end will not be able to correctly identify the >> originator of the call. >I don't know if there is anything that falls under the FCC rules. In any >event it >would be unethical and evidence of fraudulent intent if one was trying to >defraud someone in the process of doing so. Another case is in telemarketing. FCC rules require a caller-ID be present and identify a phone number where a person can request to be added to a do not call list. I am filing a complaint against a firm at present that provides a caller-ID of a non-working number! Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intel Vs AMD
what is "asterisk IMO"? Asterisk, I think you know. IMO is a common E-Mail/Newsgroup abbreviation that means In My Opinion--similar to another one IMHO (Humble). Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
>did you get your Samba issue fixed? Yes, I finally figured it out this morning. I had the domain set to match that of the web site it was serving, which was on the other side of a router, so not even on the local subnet. The DNS queries from that got Samba all messed up. Ugh! Yes, I would have gladly paid someone $150 for an hour's work to solve that problem. That assumes they could figure it out in an hour--I know any number of people who would have taken a day or two to track that one down. I can't afford that rate for that amount of time (nor does the economic model you described fit as well for an all day job). I'm a small business owner. I have a number of Linux items I'd like to off-load, but if it costs me a lot more to off-load it than to do it myself, in spite of the learning curve--well if I do it myself, at least I learned something. I'll offload PCB layout or writing DSP drivers because I can get them done for $50 an hour or less. On the other hand, if I found someone locally who knew what they were doing and charged reasonably, I'd have ongoing work for them from time to time. Yes, my charge out rates aren't the highest even in my field (which actually is more complex than Linux or networking). However, I evaluated the situation a few years ago. A friend of mine was charging twice what I was, but admitted he spent half his time marketing himself. Hmmm, he's bringing home the same per month that I am. Since I prefer engineering to marketing, I think I've got the better deal. I haven't had a down day in probably 10 years. BTW, my fees have to cover a few thousand dollars worth of assets like test equipment, computers and engineering development software. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
>This problem is only going to get worse as the so-called 'recession' bites... >fellow I.T. professionals - get used to your clients trying to weasel free >service out of you. Everything I am hearing from fellow I.T. people is that >there is no shortage of 'work' but a lot of clients are resisting paying. Not entirely. I've been trying for two years to get someone to work with my small Linux system. One guy never had time to come. I finally got someone out who was going to charge either $125 or $175 per hour (USD) depending on whether he decided it was a computer problem or a network problem (which is about twice what I charge for Embedded hardware and software development). He spent an hour here and had to go to his next appointment. My little Samba problem was beyond his ability to solve! Fortunately, he realized that he hadn't done anything and didn't charge. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] life safety system and VOIP
>The dial tone for the phone line still comes from the CO. The phone companies >>loop there copper cable in and out of the remote cabinets. Remote terminals are served by T1 or higher density carrier circuits, which can be either copper or fiber, often employing statistical multiplexing. While the DT may originate in the CO, it does so only in a data sense, not an analog POTS sense. The remote terminal actually generates the POTS analog signal, and is dependent on the life of the batteries in the box. They are good for several hours, maybe even a day, but definitely not weeks. Some RTs also have a DSLAM associated with them for DSL, but that is a separate topic and involves more batteries. >This is true, that is why most fire panels have to have 2 phone lines. Which only catches about half of the problems, assuming both come through the same cable from the same CO or RT (and, in the latter case, the same carrier circuit). If a card fails or the I & R guy opens or shorts the loop, the other line can take over. If the CO or RT crashes, or batteries die or cable gets dug through by a backhoe, guess what goes down! For serious mission critical circuits the engineer specifies two different operating companies and requires each to provide complete circuit details so he can insure that one isn't leasing lines from the other, or other scenarios that would be vulnerable to a single incident. >Time was a copper pair was supervised with a DC current from end to end, Another variation on this theme used by central alarm monitoring companies of years ago was to have the telco provide a copper loop that included a number of customer sites. Basically each site was in series. At the monitoring station was the DC power and a relay. If all was well the loop was complete and the relay operated. Each site had a mechanical interrupter--a spring wound gear mechanism that pulsed out digits by breaking the loop momentarily. When an alarm condition occurred (such as water movement in a sprinkler riser) the spring would wind down, turning the gears and pulsing opens on the loop. In some cases, this caused ink mark square waves that could be counted on paper. The pulses were similar to rotary dial pulses in groups for digits, but slower speed. They represented the ID number of the sender reporting, which identified the customer and location. Of course, if anything in the loop, any sender, any telco drop, failed, the whole set of customers was unmonitored until it was fixed--which could be a day or two in extreme cases. I was called out once to service a site that had these. The one good thing about them was the only electrical requirement was at the monitoring station. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Network architecture
>You may be able to split up some of the servers into multiple VMs -- maybe >five >servers with five VMs each. I'm not sure I see the merit in this. VMs seem to be regarded as a magic bullet (i.e. free lunch). I don't know of any case where 5 VMs can accomplish more work on one processor than simply letting the processor manage it all (except if the OS and or application can't efficiently split the task into the necessary multiple threads, which I don't think is an issue here). By definition, the total accomplished must be less with VMs, because the hypervisor will take some CPU cycles. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF not completely muted
The DADHI function is probably intended for more generalized use. Maybe for recording voicemail greetings it should not be used and a different function used instead. There is no reason why it isn't possible to backup in the recorded message and erase the blip. The detection time should approximate a known constant, and that much could be removed. I've certainly seen it done in other settings. That avoids the need for extra delay while still allows completely removing the tone. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
>It's not a matter of what I think. It's a matter of what you actually have :-) >> The third group are based on the Winbond W6692 chip. I think the >> chip was released about 10 years ago. >Which is what you have. I've known that for several months, but I believe it is considered an HFC chip. I keep responding to posts here periodically in the hope that it will connect me with someone who can help me arrive at a solution. There definitely appears to be a need (albeit of limited quantity) for a) a working US BRI solution and possibly b) a W6692 driver. >IIRC (from personal correspondence with you) you have managed to get >layer 1 working with some ISDN driver (hisax = isdn4linux? mISDN?) Linux installs a driver (hisax, I believe) for the card. Whether that constitutes "working", I don't know. I don't know what to configure or how to test at that level. F9 includes mISDN but I haven't figured enough of it out yet to know whether it can see the card. Key elements like config files and readme files aren't where the mISDN web page says they should be, so I don't have enough information to readily proceed. (Although if anybody wants to write a Zaptel/DAHDI driver for it, I'd welcome it) So would I. It isn't beyond the realm of possibility for me to write it, but it would take a very large amount of hand holding. I'm quite proficient in C. I understand the basic ideas involved in ISDN and have a working ISDN circuit, however I have very limited user knowledge of Asterisk and no past experience at the driver level or channel level and limited experience with linux. Probably my biggest weakness is that I don't have a clear picture of the levels involved--possibly because they may be historically fuzzy. If I understand correctly a DAHDI driver would cover layer 1 as well as higher layers up to what asterisk needs in a monolithic fashion. OTOH, other approaches, such as CAPI may split this up. I'm not familiar enough with what pieces cover what roles in the various possible scenarios. There are two problems that need to be solved: 1) Pieces in place to cover each layer that needs to be covered. 2) Making sure those pieces can work with US NT1 protocol. I am guessing that both have been solved by someone at some point, although probably not in the same file. I am also guessing that some modifications to existing code could accomplish this without writing a lot of code from scratch. I'm just not familiar enough with the options and available pieces to know where to look. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
>You might want to look into Cisco hardware, their WIC-1B-U cards work fine in the US, or they did 10 years ago when I last used them for VoIP. Used the WIC-1B-U is going for under $50 on eBay. An old 1600 or 1700 series router with an IOS that supports SIP wouldn't cost much either. Help me connect the dots here. I indeed see WIC-1B-U cards and 1721 routers. It looks like a pair could be purchased for probably $25. How does that fit into an Asterisk system? I can see how it would be used for 128K data, but how does Asterisk pick up and manipulate the data? Call setup? Call answering? Does the 1721 deliver VoIP data that Asterisk can process? Does Asterisk have a channel interface that can accept and use this? Thanks, Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
I discussed my installation more with Tzafrir last week. He concluded that he thinks I don't have an HFC card. I think it is somewhat a matter of semantics. As far as I have been able to determine, there are at least three general types of HFC cards. By far the most common are cards based on the "Cologne" chip--which are well supported. The second group are based on a "multichannel" chip, which seems to be fairly popular now, particularly for 2 and 4 port BRI cards where a single IC can form the basis of the card. The third group are based on the Winbond W6692 chip. I think the chip was released about 10 years ago. It has not been well supported. That is the chip my card has. It appears that no form of Zaptel or Dahdi, including publicly available patches, supports it. I'm not sure, but I think mISDN supports it, and I know my card is CAPI compliant, but that may assume a driver that may not exist for Linux. Whether either of these supports US NT1 format remains a question. Also, there is the issue of physical level signaling, which is different in the US than in Europe. I don't know if my NT1 U to S/T adapter takes care of that or not. I do know that there have been products out there that could be used in either market with only a firmware change, so maybe this is a non-issue. One of my TAs (I have several with differing feature sets and in various states of repair and (non) support) will generate a D channel transaction log, which would let me know what exchange is required with the CO. That might be compared to what the channel software does. I might be able to modify the source if necessary, although there are two or three learning curves involved for me. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
>When he examined the motherboards, both had capacitors around >the CPU that had visibly 'ballooned' A good reason to look for motherboards with either Tantalum capacitors or Organic capacitors. Its a marketing point I'm seeing these days, and as a design engineer, I can say its worth looking for. The ESR in typical aluminum electrolytics is considerably higher. These caps are at the output of a switching regulator on the CPU that is handling many amps. This creates large charging and discharging currents at hundreds of KHz rate. Any ESR (internal series resistance) turns some of this into heat. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
One relevant question that hasn't been addressed is whether just the application is crashing or the whole computer (Linux). I would second the hardware idea, with emphasis on generic hardware, especially RAM. I had a Suse 10 box that kept crashing and doing funny stuff. I ended up running an extended RAM test on it--one of those pattern sensitivity tests that takes an hour or two to run. Turned out that one of the SIMMs I had just bought and installed had a subtle problem. It would never show up on a straightforward test, but certain address ranges would fail on one or two of the exotic pattern tests. It came from a reputable vendor who does 100% testing themselves, so it was apparently subtle enough to slip through their test. They took it back and replaced it. I ran for a few weeks without the module with no crashes and when I put the replacement in everything was still fine. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail
I'm not familiar with packets specific to Asterisk, but do have some familiarity with general Ethernet traffic. The Host unreachable messages you are getting is from the protocol stack in the Linux computer, and generally means the traffic is being sent to a port that is not open--i.e. no program in the computer has requested that port to be used for listening. In this case the most likely scenario is the SIP phone has been given the wrong port for * or * is not running. It is possible there is a firewall or other security issue. I'm not as familiar with that. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
>The per-port regulators would be non-isolated. Probably feeding off >an internal 48V bus. Yes, so that will be 90 to 95% efficient, but it is fed from an isolated supply that at best will be 90%, probably less. Those numbers must be multiplied, giving 81 to 86% overall efficiency--and I am assuming best in class, which only a switch at a high price point would offer. >Well, the bulb has a peak permissible operation temperature of about >160 degrees C ... so likely no extra cooling required True in open air. However, inside a metal enclosure as I described to make it applicable to the situation, the enclosure would probably get warm enough to be a fire hazard. The light bulb might still be fine after the building burned down! :-) I've designed products and put them through UL, CSA and CE safety testing. >It'd still be a 400W PSU if it supplies 400W You can play various games with the numbers, such as choosing input versus output power to write in the spec. The label will have to indicate 500 W of power consumed, by law. No matter how one plays with the numbers, an 80% efficient supply that delivers 360 watts will consume 450 W, and turn 90 W into heat. If it is rated for 400 W output, it is probably in a portion of its range at 360 W where it is achieving near optimal efficiency and the above math would apply. In cases where half the loads drew less power and/or were not POE loads, the consumption would go down, and so would the heat. However the manufacturer can't design for that case and if they don't provide automatic fan control (which apparently most switches don't have), the fan must be designed for worst case, which in the above example is 100 W of heat. Also, at 25 to 50 % load, the efficiency will probably be lower because it was optimized for a larger load, and because some losses are fixed (not load dependent). >This whole thread is getting stupid If some increase in understanding occurs, then it isn't a waste. No it won't change the behavior of any switches, but it might help people understand why. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
>A number of 1U products use large impeller fans I've got a CPU in a 1U package with an impeller fan. It sounds like a jet taking off! Its not quiet. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
>A modern switched mode PSU ought to be more than 90% efficient, In theory, yes, in practice, not likely. It is harder to get high efficiency from an isolated supply than a non-isolated one. I get ads from IC manufacturers all the time about there 90 to 95% efficient solutions, but these are boost or buck regulators--non-isolated. State of the art in commercial practice for isolated supplies is around 80 to 85%, and typical commercial practice is more like 70 to 80%. Now you have more like 60 watts of heat radiating from this power supply. Go stick a 60 watt light bulb (incandescent) in a small metal box and see how easy it is to keep cool. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
I guess one would have to ask whether 1000 Gb is necessary. That's a lot of bandwidth. It might make sense to use it for central distribution. There are also some that have one or two 1000 Gb ports that might be appropriate for trunking and the rest 100 Mb which is probably fast enough for terminal nodes. That combination would be less power hungry. On the other hand if this is for HDTV multichannel distribution, then I retract what I said. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
>Obviously analog loops are powered by the CO, so much of the benefit of >ISDN-BRI as the first voice circuit is eroded away for a large percentage of >the installed base. That depends on the situation. For the last eight years I've been running a TA and have had to deal with that, and it isn't all that difficult (how do you say UPS). On the other hand, since this is an Asterisk mailing list, I assume anyone here interested in ISDN BRI has to address the same power reliability issues with the Asterisk server itself--I will. Once again, UPS to the rescue. Some BRI cards have a U interface, so they are as reliable as the computer they are plugged into. Mine does not, but I just plug the power supply for the S/T U translator into the same UPS that powers Asterisk. I really don't see a problem here. The adapter needs less than 10 W! I also have to deal with other loads, such as lighting and a furnace, without which I won't be talking on the phone for long. Even most of the analog phones I've been using here need local power, either because they are cordless or because they are multiline and need supervision. I really see very few cases where the need for local power for the NT1 or TA is more than a footnote. BTW, if anyone cares, my UPS works with an external battery and can power my PC, NT1 / TA, DSL, Routers and switches (with POE for access point and SIP phones) for at least four hours before I need to take more drastic measures. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
I replaced my analogue channels with digital some time back. Using a technology called VoIP... ;-) But VoIP isn't the same quality or reliability as BRI. There are extra delays that can mess up modem and FAX calls. There can be latency jitter issues as well, particularly as the most likely carrier medium is DSL or Cable modem which generally don't guarantee QOS. It certainly has its place and the cost is often lower, but it isn't apples to apples. Just how different is the US system to the European/UK one? (and why?) I may not be the best one to answer this, as I know little about what is in Europe. In the US it is packaged pretty much to replace two POTS. There are two phone numbers. There is some sharing of B channels that make it a bit more flexible than POTS. There is no ptmp. It is generally delivered and used in two wire U channel form, making multiple extensions not possible. In fact, I doubt it is often run a handset. I've been using it for about 8 years to a Terminal Adapter that converts it back to two POTS (analog). I started doing it because it could gracefully share voice and data, but then I got DSL for data. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
Yes, for starters, features like caller ID and call waiting come standard at no additional cost. In theory, it should allow two calls on either SPID, which is more flexible. Obviously in the pre-DSL days, the dynamic balance of voice and data provided very efficient bandwidth utilization. Also when I first put it in, the LEC was charging a mileage surcharge for POTS because it was coming out of a remote terminal. They weren't for ISDN, which made a BRI cheaper than two POTS. However the biggest advantage is that it is G711u to the destination. If you install Asterisk and run a POTS to an FXO, an incoming call gets digitized somewhere at the originating end, then typically stays digital to the destination CO where it gets converted back to analog. The FXO converts it back to digital. Of course there is bound to be some gain or offset change in the analog path, so the second digitization is not identical to the first. Typically you loose about one bit of resolution (out of 8) which is not too good. Try using that link with an analog modem and you will see it real fast. If your lucky, your 56K modem will do 20K, probably less. It isn't quite as obvious for voice. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
At least for this telco, ISN and BRI was a money-loser. We've spent more time trouble-shooting those connections (on behalf of the customers) than we ever made in monthly or per-minute charges. Obviously I can't speak for a telco that I know nothing about. The ability to support ISDN requires some training and test equipment, and I've seen a wide variety of approaches to managing that, some that work better than others. I do know that it seems to be thriving (as well as can be expected) here. Qwest has even used it with non-savvy customers as a method of pair gain where trunking was limited or the cost of upgrading a drop cable was high. The techs that actually work with it have proven competent and have the necessary equipment to quickly diagnose problems, often remotely from the ROC. The exception being one time when they sent a tech with no ISDN experience to a remote terminal to solve another problem and he killed an ISDN circuit in it and it took a couple of repeat visits to figure out what he did wrong. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality ofthe mailinglists
>4. Someone said "Maybe there needs to be a beginner list...". I agree >almost 100% with your oppinion but, we all know that the problem is not >(at least for now, when there are lots of documentation topics to >write) newbies questions, but bad formulated questions and people not >wanting to help them self before asking other one's help. While I understand that some might be lazy, I do think the documentation has an impact. If I go to a web site that has a well organized FAQ or a good search capability, I will try to find an answer or some tests that will help me determine the answer. However, Askerisk information is spread across so many domain names with absolutely no central organizational structure that I can determine, that I never know where to start. I've done numerous Google searches and sometimes come up with pieces, but they aren't coherent and don't always even agree with each other. The lack of organization and methods for getting to answers has been in my experience higher than just about any project I've ever worked with. If the perceived changes of success form helping one's self is about 1%, many people will start by posting a question in the hope that someone who understands the confusion and/or has been over the same path will either provide an answer or at least point them in a useful direction. I have found this list relatively good at that, but that is about the only thing that has been helpful to me so far. Relative to previous comments on how to respond, the person who said, "if you know the answer, help, otherwise ignore it", gave good advise. Some questions many can help with, others will require people with more expertise, who probably will let someone else tackle the easy ones. The one addition I would make is that it is helpful to have a few people lurking who notice something that isn't getting answered and can suggest ways to submit it that are more likely to get help, such as including contents of a config file or something. A corollary is that there are some people who just shouldn't help (or maybe even read these posts). It takes a certain amount of patience and remembering when they were at that level. They may be good at writing code or solving problems, but they may not be good at dealing with people, especially newcomers. That isn't a criticism, just a reality check. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
It seems to me that there a lot of "it ought to work or could be made to to work" associated with implementing US BRI into Asterisk I agree. I have a BRI in service and have offered a couple of times to do tests. I have an HFC card here that I intend to use with it myself, and plan to test that as soon as I can. I have gotten relatively little information and have been reluctant to proceed. However if dahdi now supports it and I can figure out how, I'll dive in as soon as I get a chance. That exercise should help get me up to speed a bit. At that point if anyone ones to send me a card of a different kind, possibly a loaner from a manufacturer, I'm willing to plug it in and try an appropriate configuration for it. I am running a production server and BRI, and it also serves web pages and will soon do E-Mail, so I need to maintain fairly good up time. OTOH, it is a small home office setting, so I don't mind being down long enough to put a card in the box and test it. It would appear to me that validating the higher level functionality and figuring out the necessary settings is at least half the battle. Once that is done, I would expect driver level issues to be more straightforward. I am wondering if there is even one site in the US using NT1 BRI at present. If there was, a set of config files might be helpful, even if they were doing it a bit differently, such as with BRIStuff. For this testing it would seem that a dahdi based approach is the clear way to go. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
I'm with you on this. A VoIP trunking solution is never going to equal a LEC PSTN solution. It may be adequate for some purposes, but I'm not about to dump my BRI for a pair of IP numbers. The trade-offs aren't worth the small cost savings for me. Just the packetized delays (not to mention internet latency) are going to degrade things somewhat. Your example about FAX is a good one. Yes, there are work-arounds, but when you have customers to keep happy, meeting their expectations in an intuitive manner is critical. OTOH, I certainly like the idea of an open source PABX doing my internal routing. I've done some stuff with commercial PABXs that few people would attempt. I've also failed to do things because the PABX OS didn't support what I wanted to do. That's one reason I once wrote my own. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailinglists
>There are far better resources out there for teaching Linux >newbies. Instead, voip-info.org attempts to provide the sorts of information >that is useful for those already familiar with Linux I can appreciate that. And I can appreciate being at the other end of the pipe, as I like to gloss over all "obvious" details when I have to write up something. I'm not suggesting that VoIP should become a Linux tutorial, but that, where possible, every line that must be typed to get to a desired end be explicitly included rather than assuming that a one sentence comment will empower the reader to type in a whole page of bash stuff. >You could certainly compare and contrast the documentation for other >large daemon applications I would concur with your thoughts here. The terse style is endemic to everything Linux. I bought a commercial Linux app recently and it didn't even have a single word about installation. Turns out there was no configuration, so you could just drop it into a directory and make a shortcut icon if you were using Gnome or something. But at least a line stating that would have saved the author an E-Mail exchange..Oh yes, and then there was the library it needed that wasn't in the distro. It would have been useful if the Readme mentioned that. Anyway, you get the point. >It's certainly instructive that the continuing advances in >open source browser technology was what spurred Microsoft to once again >invest time into its own browser True, but I can open IE and use it and then open Firefox and intuitively know what to do. It doesn't say you can use Z or -M or --Query all to do the same thing, (possibly with identical parameters, or possibly with parameters formatted differently) but -M only works on Fedora and --Query only works on Suse (contrived by true to life examples). It is this sort of thing that makes the Linux learning curve steep and makes it challenging to provide detailed instructions for something like installing a package. Based on a number of conversations over the last year or two, I have become convinced that those for whom these command automatically flow off of their fingertips are mostly clueless as to how unintuitive some of this stuff is when first encountered. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
Thanks for engaging with me on this. I picked up the book and I see what you mean about Appendix B. I had under-appreciated it probably because of a paradigm shift I need to make. I think you meant Appendix E rather than F for dialplan. I still am not quite on the same page with you, though. There are a lot of commands that aren't function calls that go into various config files. The most basic and obvious one is exten There must be a hundred of these and I don't know where they are listed with all acceptable parameters and ranges and what they do and why. There are examples to get one started, but I don't think I can put my hands on even a definitive definition of exten. Am I making any sense? Maybe these are called variables or something. I'm scared to even look for the setting for an NT1 ISDN BRI, which is the mountain I have to climb next. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
>Rather, can you tell me who claims to support what hardware, so I can confirm? I don't have any notes on what I did. It was a bunch of Google searches. I seem to recall that Digium themselves made a two port BRI that would work. Eicon has some well respected products that I am pretty sure are supported for NT1 (They are based in Canada). Zyxel, I think does, also. I didn't go that way because most were two or more ports and I only needed one and couldn't justify the $200 to $400 cost. >I have no time for digging and experimenting, unfortunately. I expect that >time will be >taken up entirely trying to install and configure asterisk. That translates fairly straightforwardly into, "you have no time for Asterisk or Linux, its underlying operating system. I can't imagine going down this path in any form without setting aside weeks of time for experimentation. I've invested at least two person weeks to get Linux up and running and a small demo Asterisk install with no trunks and two SIP phones. I think the payoff will be worth it, but the learning curve is steep. If you are already capable of installing Linux from source code in your sleep, you mileage may very significantly. >Okay, these are various driver protocols in the kernel/user space, right? Yes, Dahdi is tightly linked to Asterisk, the others are more loosely coupled. >Since I haven't purchased anything, should I be analysing the rate and support >for >development of these different protocols and informing my decision that >way? Based on the related posts today, it sounds like you should be looking at what dahdi can do and what drivers the board makers provide. The other options are possible paths to a solution but no where close to turn-key. >any solution that doesn't lease equipment from the provider conflicts to their >bottom-line >interest, so I don't expect any help from them. Am I wrong in >that? I've been getting ISDN from QWEST for about eight years now at a combination of four locations. The company has made a huge amount of progress in operations over that time. The last two installs went off pretty much without a hitch. The sales people tend to be weak on ISDN, and the I & R people are variable. My best result has been with the guy in the CO that works ISDN circuits from the inside. In our office, he's sharp and quite willing to give out his phone number to anyone who doesn't come across as an idiot for follow-up questions. I wouldn't expect them to tell me how to configure Asterisk, but if they have a problem, or sales didn't write up the configuration correctly, he's been able to answer my question and change the CO programming on the spot for me. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
>It actually does contain references of all applicaitons, CLI commands, and >such. Where? I saw some examples, but I've never found an organized list of commands. I'd love it. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
>I wonder if BRI would have gotten traction if it offered PRI functionality I can't say for sure, and don't even know the differences in functionality, but you may be right. When I last ordered DID I couldn't justify PRI so brought it in as analog. At that point in time and with that LEC PRI wasn't cost effective, even if I filled it up. I could have filled a 24B but it would have left my entire facility at the mercy of a single circuit, which isn't very smart. To me the thing that did the most to insure the demise of BRI in the US was the insane pricing. Most LECs saw it as a large cash cow and priced it with large margins (keep in mind it is cheaper for a CO to export two LDNs on a BRI than on two POTS, but most priced a BRI at about 4x a POTS). Most LECs only offered it as measured service with stiff per minute charges even on local calls. Many charged stiff rates for connect time in data mode (in fact some would get around this by treating data as voice). Their greed backfired, and of course when DSL offered more bandwidth for considerably less money, the bottom fell out of the data side. The only LEC I knew that didn't go down this path was QWEST. They priced it flat rate at rates that were competitive with POTS and I leased them for every site I managed. They offered higher quality and more features than POTS. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
>I'm impressed that you picked up 6502 assembly out of an even larger >"vaccum" considering there was no 'net back then to help at all. Did >you install a PBX on an Atari? No, I interfaced a Rockwell AIM to a 300 station Philips electromechanical PABX (designed and built about 100 interface cards, including DTMF receivers) and then wrote all the call processing code. The Rockwell AIM did come with manuals that completely documented both the hardware interface and the instruction set. In the days before the 'net, such paperwork was mandatory. >it just requires diligence, patience, I'm trying. >It certainly helps to be Unix inclined, Unix was barely out of Bell Labs when I got my CS degree and we never saw it, so I am at a disadvantage. I have worked a bit with a couple of Unix installations since and do have a computer running Fedora 9 and one that is supposed to be running Fedora 10 64 bit if I can ever get past a kernel bug, so I am trying to come up to speed. I am a lot more familiar with what to do after the reset vector on an 80186, or the inner workings of a protocol stack. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
>YMMV. Mine certainly did. For the better. My comments were more negative than I intended. My installation is "worthless" at this point because it is only a cookbook example and I haven't tried to modify it to meet my needs. I didn't intend to imply that Asterisk is worthless, just that I've only gotten to the point of a trivial demo. My main concern is that the documentation isn't for the faint of heart. If one doesn't devote many hours, on a regular, ongoing basis, they may never get to the point of understanding it enough to apply it to a real-world situation. The more I explore and the more feedback I get, the more I find is there. I just got a very nice posting from Tzafir showing me a web domain I didn't even know existed. Not surprising, it is a lot like Linux--everyone has there own idea of what is needed and how it should be done, so it becomes a monster that is hard to get a handle on. From what I've seen so far, the commands far exceed any commercial PABX I've ever used or evaluated. It is very powerful, but the learning curve is immense, and I'm both a CS professional and a telephony professional. I'm not abandoning it by any means, but am frustrated at even where to jump in. I excitedly bought the O-Reily book, only to find that for all 1000 pages, it never provided anything that could be considered a reference manual and that its tutorials weren't even a good fit to my needs. It did get me two SIP phones talking to each other and to a softphone, but only after hours of experimenting with SIP phone settings and contacts with the manufacturers (who knew even less about VoIP). I think part of the problem is that the only people who know enough about * to address the documentation problems are busy either developing hardware and software for it or using it to run their businesses and don't have time to address the documentation problem, which is understandable. Also, once a person gets to that level of knowledge, its easy to forget how little a newcomer knows and leave out a lot of necessary details. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
Thanks for the reply. I have looked at the links you provided and I think they will be useful. I may have some issues with drivers for the HFC, but I guess I won't know until I try it. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
>To the best of my understanding, latest Asterisk should support it >through chan_dahdi . No need for extra bristuff or whatever. But this >needs some testing. Any chance I could get some information on how to set it up and use it (keeping in mind that I have limited Asterisk experience and no experience with zaptel or mISDN or BRIstuff)? I have an NT1 BRI here that is up and running and an HFC card and an F9 Linux with *1.6. The BRI can easily be moved to the HFC card for testing. I installed 1.6 from an F9 Yum distro, but I could probably figure out how to download and make the latest from source if necessary. I did the F9 installation and run it, but I'm not real strong on Linux. I mostly do 80186 embedded development in C and assembly. >BRI there behaves very much like PRI. No extra ptmp complications as in the >rest of the >world. I never dealt with PRI, so I can't say for sure, but you are correct that it lacks ptmp, etc. >It seems, though you may need to do some custom wiring. AFAIK, only that it is delivered in U form and some cards (including mine) require S/T, but I already have a converter for that. I am anxious to get my BRI into Asterisk, I just have been overwhelmed at the terse mISDN documentation and haven't jumped in. Having it in dahdi would at least mean one less separate piece to mess with. If I can get some documentation or a bit of handholding, I will take the plunge, report any problems and document the end result (assuming it is successful and worth documenting). Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
I'm in the same boat and have been looking at this for several months, but haven't actually jumped in, hands-on, yet. No, I don't think the situation is as dismal as you paint it, although the lack of appropriate marketing for BRI in the US has all but killed it here, making it relatively unattractive to vendors. I have been advised of several cards that support it, but they were two or four port cards that were serious overkill for my application, and expectedly pricy as a result. I am trying to use an HFC card, and have been advised that it is possible, but will take some digging and experimenting. As near as I have been able to figure out, mISDN is the appropriate place to begin, but I haven't tried to sort it out, yet. There are a couple of other paths that might lead to a solution, but they seem to be less supported than mISDN at present (if that is even possible). It sounds like Dahdi is moving towards that, but I don't think its quite there, yet. Qwest is one of the few providers I know of that prices BRI reasonably. I have one up and running on a TA and use it for my voice service. As soon as I can figure out mISDN, I plan to move it to Asterisk. (I'll keep a TA around for backup). Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
>It seems to me that everything one may want to know would be contained >on voip-info.org My own experience is that it covers a very broad spectrum (far broader than Asterisk) and in a rather terse manner. I have spent an hour or two at a time pouring over a topic there and come away little more enlightened than when I started. Most people who know enough to create useful entries there, assume too much of the reader. They assume that everyone reading the post works with Linux 40 hours a week at the command line level, and only needs a few VoIP clues to take an idea and run with it. A better assumption would be that they know how to log on. It shouldn't even be assumed that they know the difference between suand su - I realize that this is challenging because different distros do things different ways. That is another topic of its own, but is also one of the banes of Linux that is hurting its usability considerably. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
>New to Aserisk 1.6 and find the 'installation tutorials' seem low to non >existent. I first looked at * about four months ago and rapidly came to the same conclusion. Even with the O-Reilly book, which I purchased in paper, although it is freely downloadable, I feel there is a huge dearth of information. As I have become a bit involved, I find there is more than meets the eye, but it is spread across the entire internet! So far I am not aware of anything that fits any of three categories I feel are essential: 1. A good tutorial with enough detail to allow a person with a CS degree, years of telephony experience and limited Linux experience (myself) to install and configure a reasonable * system (something more complex that an FXO or two and a couple of SIP phones. 2. A reference guide that lists all commands and options with explanations of why they are useful and how to use them. Even the book doesn't attempt to touch this one. Such a reference needs to include things like Dahdi and other pieces that aren't strictly part of * but without which few installations could exist. 3. A decent cross-reference that can quickly allow someone to find the scattered information available on the web. Even this mailing list is so hopelessly linear in nature compared to most other newsgroups I am involved in as to be almost useless to me. My conclusion after installing a worthless * demo (that actually does allow two SIPs to talk to each other) is that Asterisk is not of any value to anyone other than a person who makes a full time career out of running Asterisk systems. I've installed and maintained several traditional PABXs and even wrote the control firmware (in 6502 assembly) for one, with sizes from 6 stations to 300 stations, including things like DID. It was kindergarten compared to Asterisk, and primarily because of the huge information vacuum. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
>making sure to patch any holes through which the hacker might have come In my case, I had been getting regular attacks through SSH for months, probably 100 a day (bots). Apparently after nine months of this, someone stumbled on to my password which regrettably was composed of two dictionary words with no special characters, making it susceptible to dictionary search. When I re-installed, I put SSH on a non-standard port and haven't had a single attempt to attack the system since. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
The guy who hacked me didn't seem too concerned about not being noticed. The replacement ps would not allow me to kill any processes (including the ones he was running). There was enough log information left that I could trace the intrusion and even the ISPs hub it came from and I reported it to the ISP, although I don't know of they followed it up. If he was clever, he was going through some innocent person's computer already, which would have pretty well covered his tracks. In this case, there were only three or four passwords in the system, so I wasn't too worried about that. Tripwire would be fine, if it had a baseline, but I don't think its any good after the fact. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
There have been a number of answers provided. The one that was given to me when I encountered this same problem was to boot a live CD, mount the root file system and delete the password file which would force your normal distro boot to request a new root password next time. HOWEVER, the big deal here is that the most likely cause is the server being hacked. I got hacked a few months ago. Step 1 was log in as root. Step 2 was change the root password. Step 3 was replace a few key executables like ps so I couldn't do administrative tasks. Step 4 was launch a denial of service attack against someone. That is when I discovered the problem, because it ate up all my DSL bandwidth. The problem is that you don't know exactly what files have been changed and if they have left a trap door or something. You could fix the root password, and even discover and restore a few key files, only to find it hacked 5 minutes later because you didn't know everything that had been altered. For that reason, few people will put a system back on line after the root password has been compromised. Re-installation is the only safe way. If some of your directories like /home and /user have separate mount points, they don't have to get wiped out in the process. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not Dialing 9
>> In North America: >> 0 is the intra-lata operator >> 00 is the inter-lata operator >> 0+ will be an operator assisted call >AFAIK this is not correct at least here at east coast (MA, NY, NH...) >1 is a national call (local or long distance) >011 is international call >Some providers allow you to dial without the prefix in their network In some places 0+ is intra-lata operator assisted and 00+ inter-lata operator assisted. In other places (like CO) we don't have LATAs. The NA number plan has been and still is much more uniform than in many other countries. However, given the variety of unregulated providers plus the breakup of the Bell system, it has diverged some. The basic NPA-NXX- scheme has remained (so far) but details like 1 being optional on Cell phones, whether 1 means "area code to follow" or "this is a toll call", whether local calls and/or calls within an area code require 7 or 10 digits is up to the LEC in each case--its a bit confusing. Then there are LATAs (or not) and some places which have mileage based charges that aren't quite long distance, but aren't local, either (ZUM). I managed a PBX in the 805 area code where the LEC provided 7 digit local dialing, 8 digit toll dialing within the A/C and 11 digit toll dialing outside the area code. Except that two prefixes from the 818 (originally 213) A/C was also local and you had to dial 11 digits for that (1+) even though it was local. Furthermore, it was run by a different LEC that had different dialing rules, so making the call in the opposite direction had different rules (10 digit for local in the area code, if I recall). A bit confusing for someone who lived in one community and worked in the other and made calls both directions between the two. We're pretty well off here in CO. We do have an overlay area code (303 and 720) so everything is 10 digits, but everything in those two area codes is local, there are no ZUMs or LATAs. The only abnormality is that there are a couple of other areas codes in the state that have different tariff rules than out of state, but it doesn't affect dialing, only cost and provider. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not Dialing 9
I set up a couple of PABXs this way 25 years ago. It was a little simpler then because there was a more uniform number plan in the US back then, although most of the industry people I talked to though I was totally crazy. It was a 300 station 3 digit extensions system. It worked well for a number of years. As was previously mentioned briefly, early PABXs not only didn't have pattern matching, many couldn't do store and forward dialing, so 9 literally connected you to a trunk where you got a second dial tone from the CO before you could do more dialing. That's this history of it. Where I live now in Colorado we have 1+ for long distance and 10 digits for the two local area codes. Other areas are different. Since the two local area codes are 303 and 720, there are four leading sequences possible 1+ for toll 303+ for local 720+ for local 0+ for anything operator or credit card related. For my SOHO setup, I opted to used two digit extensions of the form 2+ If I run out, I will use 4+ The one I did 25 years ago was more complicated because the CO allowed (so people were used to) 7 digit dialing. LD could be 1+ seven or 10 digits. I basically avoided local prefixes and hoped the CO didn't throw me too many curve balls. The biggest curve ball was when the changed the meaning of 1 from long distance (7 or 10) to area code prefix. Suddenly what people had to at home created conflicts between non-local prefixes in the area code and extension numbers. We just left the old rules in effect, as most people were dialing either local or out of area code anyway. BTW, I'm not aware of any shift towards 8, except that the Hotel/Motel industry has been using both 8 and 9 to differentiate between local and long distance. It makes programming the switch much simpler. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A method to determine PSTN Call Provider?
With number portability? I would think that would require a very large and dynamic database that would get obsolete rather quickly. I guess somewhere in the deep innards of the network that database has to exist, but are mere mortals allowed to access it? Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increase DTMF Tone Duration
>The problem is simply the duration is too short (120ms), and the remote IVR >>seems to not detect them That sounds like an IVR issue. I've worked on some traditional PABXs and even designed some DTMF receivers. Any decent DTMF receiver should be able to reliably decode 80 ms tones, and a really good one can decode 40 ms. 120 ms should be a very generous duration. I shipped 80 ms duration to COs 20 years ago. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Message 0841984
What's scary is if the account used was a legitimate user and was hijacked, and either spoofed or the computer has a trojan that gives them control. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC Single port Cards
I have a small Asterisk system here that I'm wanting to set up for a SOHO using one HFC for a US NT-1 ISDN. I haven't had a lot of time to spend on it so far. I have a couple of SIP phones talking through it, but not even a decent dial plan, yet. But I keep monitoring this group and occasionally asking questions. I'm not sure what route to go yet, and my lack of strong Linux experience makes for more pitfalls than some might have. If I put some pieces together with something missing, I probably wouldn't know how to figure out what was missing. The opinion I have formed from comments here and reading elsewhere is that mISDN is probably the best way to go, but the things I have read either get into technical details that probably don't apply to me and are hard to understand, or assume the reader knows a lot more details than I do and therefore paints in overly broad brush strokes--or both at the same time. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk latency
Depends on how much latency. The packetization of voice data (and associated digitizing, transcoding, etc) introduces some latency. Smaller packet size can reduce this, but at the expense of needing more packets which eats up more CPU time, etc. Also the jitter buffer size makes a significant difference. For a PBX (LAN) application this can be quite small, as network processing is fairly predictable. For stuff going over the internet it needs to be larger. I have a small demo setup I'm experimenting with that only has a couple of SIP phones. They are in the same room and the delay is was very annoying. I made the jitter buffers smaller and it helped. With good echo cancellation and more realistic physical separation this isn't really a problem. If it is network based, you will see it on a ping. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
I guess there is a variety of opinions on this, some of which relates to the tools a person is using. The absolutely most offensive thing to me in a post is to have to scroll through a bunch of copied original material that I've already read six times to get to the new part. My own preference is not to quote anything other than a short phrase snippet that is directly being replied to or failing that, at least put the original after the new material for those who might want it. I realize that not everyone sees it that way, but maybe it throws a different perspective in the mix. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Re: Threading etc.
Thanks. I have them going to their own folder now which is a great help. It doesn't appear that OE can organize them in the folder, though. I does a good job in a news folder, but doesn't seem to have that ability in a mail folder. Nor does it create a usable To field for replies. I still have to change that every time. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another
>Never heard of that mailer. You might try using Kontact under Kubuntu, as it >has reasonable defaults. Outlook Express, the default mail handler on Windows XP. I only have one Linux computer at present and it is dedicated to Asterisk and file backups and web serving, etc., so whatever I use has to work under Windows (I use a lot of aps and exchange files with customers that require Windows to be my normal OS). My real question was more how this is supposed to work. I've been reading posts and sending a few, and seem to be functioning OK, but I don't know if I'm doing anything right, and using the tools I'm using it is 10 to 1 more chaotic and less useful than anything else I've ever done, so I assume other know a better way. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another
>Try to use a decent mailer that does not break threading. This is an opportunity for me to ask a question regarding this mailing list. I've worked with several other groups using a variety of communications techniques from Web based to news reader based, but never anything like this. Due to my lack of experience (and/or wrong tools) this looks totally like chaos. Asterisk messages come in with no apparent thread ordering other than noticing the subject line. They are all mixed up with my general E-Main (and spam, which they exceed in volume). I can't reply to them, as the To line generated isn't viable. Is there some organizational structure I am missing here? I am using XP and OE 6 for E-Mail, which may not be that friendly towards this type of situation. It does have a very good newsgroup reader that I regularly use for another group I am part of, and I would be much better served by that than by having it all land in my general in-box at the rate of 100 a day, all mixed up with my personal and business correspondence. Thanks, Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CPU Usage
>When system is busy, asterisk uses 99.9% cpu. >I want asterisk to use more 100% cpu to process more calls. Am I in a time warp? Is this April 1 in disguise? The difference between 99.9% and 100% is not discernable. The other .1% is probably used determining that asterisk is using the rest. Even if it were, this would be an OS issue. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities
Yet another option is a commercial system with in-house staff. I used to maintain a NEC (NEAX 2400) for many years. I went to factory training and had total responsibility for it. Some manufacturers discourage or prevent this, but others are open to it. There are also 3rd party organizations (such as Source) that can supply parts and even expertise for those going that direction. Whether the result would be higher availability than Asterisk, I don't know. Given I'm both a telco guy and a computer guru (CS degree) I'd probably go the Asterisk route myself, because its open and I would have more control. Wilton >and bug fixes than any commercial product sold in the intra-industrial channel ... and they won't charge you a $30,000 license fee for the upgrade.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)
> In fact, the amount of time it takes to fetch memory from a cache miss > can easily ruin the single element lookup performance in a look up > table. That's a good observation I hadn't considered. No, none of the CPUs I work with cache anything other than the next few instructions. But, like you say, that isn't true of just about any CPU Asterisk is likely to be run on. It used to be that code optimization was fairly straightforward. But between caches and pipelines and multiple execution units, etc. It is becoming a very complex science of its own, and as Benny stated, often very CPU (or even main board) dependent. I guess the bottom line here was the comment about 1.4 X 0 is 0. These two codecs don't contribute significantly to CPU utilization any way they are used. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)
>For those of you interested in a slightly longer discussion here, there is >discussion >(Nov 14) Thanks for the link. I just listened to the segment. It sounded like the problem was going between linear and ulaw. As has been previously mentioned, this may well be due to a broken algorithm that has been widely published (as has the fix). It is absolutely inexcusable for any errors to occurs simply going in and out of ulaw. ulaw is a lossy compression, but the inverse function should always return a point near the middle of the range that maps to a given ulaw number, such that barring significant errors in the analog domain, a subsequent encoding would generate the same ulaw value. Again, both ulaw and alaw (and their inverses) should be simple table lookups, so getting the table right has absolutely nothing to do with speed. Also, in response to some questions asked both here and on the conference call, neither ulaw or alaw involve information about more than one sample at a time, so there is absolutely no need of buffering or anything else that would impact latency of the call. In a SIP environment, it would probably be processed packet at a time, but the packet size is determined entirely by other considerations, not the codec algorithm. I'm not familiar with either the old or new code, and there may be some general programming areas where code was cleaned up, and the cleaner code was less efficient. However, there is absolutely no reason, assuming a LUT implementation, that the transcoding fix should have any impact on speed. The proper implementation is a single array element lookup. Where a possible issue does exist (and it is intractable) is going between ulaw and alaw. Both are lossy compression yielding an 8 bit output, but using slightly different mapping. That means that there cannot be a unique mapping from one to the other. For some values a given ulaw number has more than one possible alaw equivalent and for other values a given alaw number has more than one possible ulaw equivalent. That means that each transcoding between alaw and ulaw (or reverse) causes a slight loss of information. Best case scenario would be that after two transcoding, standardized values are used such that no further degradation occurs. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)
>It wouldn't hurt for you to do a code review on them, I'd better get more up to speed on * in general first. It would be interesting to compare them to my code. However, I don't have a useful * installation here, yet--I'm working on it. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)
I'm a bit puzzled, also, having implemented ulaw and alaw in an embedded application. Each can be done with a 16 Kbyte table in about 0 time with no errors. There are probably tricks that will cut the table down by 2 or 4 X for a small cost in CPU cycles. The inverse requires 256 16 bit words. I thought ulaw and alaw were pretty much no brainers. I don't know of any gottchas. Why anyone with more that a few K bytes of total system memory would even consider anything other than a lookup table is beyond me. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend Wireless IP Phone
>"naturally wooded" does not bode well for WiFi True, and it's even worse for the 5.6 GHz stuff that most of DECT is using these days! The marketing departments have everyone convinced that bigger frequency numbers are better. For most real-world environments the exact opposite is true. The only advantage I know of for higher frequency is more available bandwidth and less congestion. >If the road is fairly straight More like a boomerang going around the highest part of the hill. >You may be better off with something that uses lower frequencies True. I have an Engenius high power 900 MHz unit that covered the property fairly well and about half of the road, and that with the base station in the walk out basement at about the lowest point on the property. Unfortunately it went up in smoke one day. Also in a POTS environment it could only work with one line. We'll see what this setup does with appropriate location of the base. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend Wireless IP Phone
>It's definitely possible to make wifi work for half a mile and you don't even >need a >600mw transmitter to do that - however, wifi is all about receive >strength, and so >you are unlikely to get a significantly better coverage with >a high power hotspot >which is suboptimally placed. If you do go that route >then getting the antennas into a >location where 90% of the signal isn't >already killed going through walls before it has >to travel some distance is >the trick. Probably also consider a repeater of some sort >rather than just >one high power device Good points. I got an access point instead of a router specifically so I could locate it in the best position. IMO Wi-Fi routers are dumb by definition because where you want a router is probably NOT anywhere close to the best point for the Wi-Fi part. This unit has a particularly sensitive receiver to compliment the higher power. It would have been nice it it had MIMO, too, as that always helps. Repeaters would be a challenge in this case because most of the property is natural wooded (so no power or protection) and I'm trying to cover a road by only own property at one end. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962 & Asterisk
>The timezone only tells the system with what offset to show >the time when asked for "local time". >Sadly some operating systems have this strange concept that changing a >time zone means changing the system clock itself. This makes it a huge >change indeed. Agreed. The firmware I design works the same way--everything internal is in UTC. Any application that must deal with multiple time zones by virtue of market distribution or because it shares time over a network, etc. should use UTC internally and only translate to local time. Using a scheme such as *nix does of an integer rather than broken down field makes the translation trivial. The hard part is deciding how to determine the translation, whether to use hard coded rules, intelligent observation or manual setup. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962 & Asterisk
>The linksys phones annoy me because they cannot implement southern >hemisphere DST properly. I was shocked the first time I had to write firmware for an international project. Not only is there the southern hemisphere issue of opposite seasons, but just about anyone in the world with a legislative body has to prove their independence from everyone else by defining the dates a bit differently (not to mention time zones that differ by 15 or 30 minutes). Then the US came along and changed their rules after a million products already had them hard coded in silicon! It's a mess. I just wish we'd all forget about it entirely. Its a way to force people who don't like to get up early to do so anyway. A number of studies have been done on the increase in accidents and reduced worker productivity for a week or two after a change. The recent US change was supposed to save energy, but I suspect if one did a study, they would find that businesses just extended their hours to accommodate a diversity of people, thus increasing their energy consumption! Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] twice normal beep before busy tone ??
If it is 300 ms, that is way to long. I don't know any CO grade receiver that can't decode in 80 ms and some can do 40. There is also a similar size gap between digits. Is there an option to start dialing as soon as enough digits are collected to guarantee a unique route? That has been the norm in traditional PABXs for 20 or 30 years, and combined with 80 ms duration, it can generally finish by the time the user has entered the last digit. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend Wireless IP Phone
Wi-Fi SIP phones aren't limited to hot spots. I am in the process of setting up asterisk for SOHO. At present I'm not even using VoIP trunking, only LAN to stns and I intend to use Wi-Fi instead of analog cordless phone. I got the Engenius one, and it works, but I haven't played with it much. I was disappointed that it only has a single line appearance, as part of my reason for going SIP was to allow the same features like say my 941. I also got their 600 mw access point, but haven't had time to try it. My goal is to cover out 3 acre property and the 1/2 mile road to the mailbox, including mountainous terrain. Maybe I'll share more when I actually get it all put together. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phishing attempt
I wasn't on the list to get the survey, but have been reading the thread. This is an important issue for anyone involved in e-commerce, with all the phishing and spam circulating. I am wondering if a couple of simple steps might mitigate concern: 1. Place a web page on a Digium server with a recognized URL that explains the survey and documents the domain name used in the E-Mail. 2. Place a link to that page near the beginning of the E-Mail in plain text form so it is easy to see and validate. This would allow crosschecking. This would even pre-empt someone trying to fake the E-Mail, as the URL listed on the Digium web page would not match the link in the fake E-Mail. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN - BRI
Subsequent to some previous E-Mails, I've been trying to dig into the ISDN - BRI situation a bit more. I have determined that I have a HFC card with Winbond chip, but I'm not sure what combination of drivers is best or usable. zaphfc is out because it only supports the cologne chip. misdn is a possibility. I haven't determined if it supports the card natively, or needs a card specific driver under it. capi is a possibility, but again, I don't know what driver, if any needs to be under it. capi can support misdn under it, but I don't know if this is an advantage or not, and again whether a card driver needs to be under misdn libpri 1.4.4 is supposed to work, provided you unpatch the bad patch in the source and compile it--again, I'm not sure what driver, if any needs to be under it. F9 detected the card and loaded some sort of driver support for it, but I don't know if that covers the lower levels appropriately. Can anyone provide more specific information or suggest which of these approaches is most likely to work, most likely to be stable, or supported for the future, or support the most features? It would appear there are at least four possible ways that might work, but I can't determine which is best. Thanks, Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
>copper has nothing to do with it. You're talking about conventional PSTN >(circuit >switched) technologies. Even more succinctly, it is the difference between streaming and packetized data. Circuit switched isn't even a necessary qualifier. Dedicated bandwidth with minimal buffering is (i.e. fixed path or timeslot allocation). Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] whisper time remaining
>From a technology standpoint, if I'm not mistaken, this would require some >sort of conferencing because it is mixing two digital audio streams. Call it >what you like, but it has to have extra resources. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
Thanks Brendan for the explanation. There is one other idea that struck me, but again, I don't know if it has any merit. My thinking is to keep FAX as FAX and electronic as electronic, rather than introducing a new hybrid approach. Obviously Entering FAX from an electronic source is as old as the FAX modem, and Exiting it electronically is as old as E-FAX, not to mention other alternatives. Is it feasible to simply specify the codec as ulaw or alaw (depending on jurisdiction, I forgot the g numbers) for calls originating from the FXS or whatever the FAX is coming from? Obviously, the bandwidth would be higher in that case, but you can't get around the laws of physics. Yes it is lossy compression, still, but it is the simple, predictable form of lossy compression that the modem in every FAX machine already is programmed to cope with. The only problems I can see would be if the provider who handles the call refuses to accept that codec, or transcodes it to something else. I don't know the likelihood of either of these. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
> "People can't figure out e-mail as it is, they aren't going to figure out how > to fax via >e-mail.". I can understand people saying that. Myself, I'd take E-Mail any day. I've been messing with FAX at various facilities for years, and have found it unreliable, as have most people I talk to. Nobody knows of the FAX actually went through, and if it did, whether the result was readable on the other end, not to mention if the wrong person grabbed it, or accidentally threw it away with some SPAM FAX. Oh well, each to his own. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
I've been following this thread and trying to sort out what is wanted, what is available, and why. Comments to the following would be appreciated and might be useful to others. 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. 2. Why would anyone terminate a FAX call coming through Asterisk in a FAX machine? Isn't there a way to capture it electronically? If so, it seems that putting the electronic documents in a queue where people can open them, save them, and if they wish, print them would be much more useful (and planet friendly, since a lot aren't worth putting on paper). IMHO, there are only three realistic needs: A. Electronic end to end document transfer which is best done with E-Mail and not telephony. B. Receipt of FAX from outside (old school) sources, which is best done electronically. C. Generation of FAX to outside (old school) destination, which could be done either electronically or in the traditional manner. If end to end FAX is desired, is there any reason why Asterisk should treat it any differently than any other call? The FAX machines on each end generate and decode the information, VoIP is simply an audio channel through which is passes. I don't know what T38 defines or implies, but if it is anything other than how to electronically decode a voice call that happens to contain FAX information (rather than passing it on to a real FAX machine) then I'm not sure what use it is. It would seem to me that the OP needs a way to electronically capture calls that turn out to be FAXes. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-users] +heartbeat
>My guess is that if you had two NICs on the same subnet with >different IPs the kernel route table and ARP cache would get pretty >confused. This seems so incredibly broken to me I've never tried That was my guess and point to begin with. I was not aware of or thinking about bonding. Without something out of the ordinary in the protocol stack, there is no way to determine which NIC the OS will use for a given destination IP, since either can get there. That is why the hi-rel stuff I do has two parallel LANS with different subnets. It places more of the burden on the program to know how to do backup, but I control the code in the projects I'm doing, so its not a problem. Wilton___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-users] asterisk +heartbeat
> having two NICs on the same subnet I'm trying to wrap my brain around that in the larger network picture. Two NICs in the same subnet (presumably on the same computer) would have access to the same other devices. This could potentially increase bandwidth (maybe?) and offer redundancy (if NICS, wiring or switches were the biggest source of failure). I'm not sure how the OS would decide which one to use for a given packet, or if an application (such as Asterisk) could determine which one to use. I can see potential problems with addressing, as other devices could send to one, and would definitely not know what to do with a reply from the other, etc. I'm not sure this would be an Asterisk bug. Without some concept of what I am missing here, I would consider it a cockpit error on system setup. The only reason I can think of for having two NICs in a computer would be using it as a router--in which case they wouldn't be on the same subnet. (OK I've done it before for redundant paths, but again, the paths should be on different subnets, otherwise how does one tell the OS which path was intended?) Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
>There's a HFC-S winbond card. How does that card show on lspci? Network controller: Dynalink IS64PH [0675:1702] ISDN Adapter kernel modules: hisax Does that tell you anything useful? Do you want more details? Would you like to borrow one for a while (I have two)? Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
>Why didn't BRI catch-on in the US? I a word--greed. It arrived shortly after divestiture when there was a lot of competition in the market and a dozen independent regional telcos. Apparently they saw a huge cash cow for this data service and yet another competitive advantage to proprietary implementation details. I used to live in GTE territory (now Verizon) and they were charging 2 cents per minute of connect time! This was on top of monthly fees that were unreasonable to begin with. I don't know if they have all caught on now and fixed it, but its too late now because most of the US has DSL which is 10x the data rate and supports VoIP for voice. I'm fortunate here that Qwest has offered it for at least 10 years at rates comparable to two POTS lines and no per minute charges. When I started using it, in my situation I got more features for less money than two POTS lines, which I would have needed instead. The irony is that the direct cost (equipment) of ISDN is less per B channel than POTS. Anyone who has ever compared the cost of digital versus analog station cards for a PABX, knows painfully the cost of supporting A/D conversion, 90 V 20 Hz ringing and even DTMF registers. If they had wanted to the telcos could have made ISDN cheaper than POTS, still made money and moved technology forward in the process. But to illustrate the mentality, GTE, who I mentioned above was serving the affluent community I was in with stepper switches in the CO until well into the 1980s! They put tone to pulse converters in front of them (initially for an extra fee) so they could support DTMF. Ironically they only put 600 ms interdigit time in the converters and the steppers could take up to 800 ms to find a link for the next digit, so the call failure rate ran 10 - 50%! I ran a PABX at the time and we did our own tone to pulse conversion just to avoid that. With 800 ms interdigit time we had at least 99% completion! Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
The card I have has no name but is based on the Winbond W6692CF chip and ships with RVS, which I think is for Windows and of no use to me. I'm not sure about whether it is supported by DAHDI or not. Wilton___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
>I have done this. Good >Why BRIs exist in the US is beyond me. I'm not sure why you say that. It is the only way I know of two get two digital voice grade circuits at prices competitive with POTS. The better question is why the LECs used such poor judgment when they introduced this. Most were charging outrageous prices that had no technical justification and per minute usage fees. They destroyed the market before it got off the ground. I don't use it for data any more now that I have DSL, but it is still a very viable voice channel. If you can, don't go with BRI. Why? (Not that I don't already have it and have been using it for seven years.) Who is the carrier. Qwest (formerly US West or US Worst as some used to call it). So do you have some information about what you did or where to get configuration information? Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
>With ISDN, the conversion is done in your phone Exactly. Or in the case of Asterisk, it is a 4 wire digital right into the switch--no degradation. Even converting back and forth between analog and digital multiple times compromises quality. Try doing a dial-up modem across such a path. The best you will get is 20 - 30 K. >IF you can get a PRI-line for the same price. Not to mention that the interfaces for PRI are about five times as expensive. I'm not sure why. It doesn't seem like it ought to take a lot of electronics to break down the bit stream. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
