Re: [Asterisk-Users] Recording voice messages in mp3 format
[EMAIL PROTECTED] wrote on 11/19/2005 02:04:54 AM: Have you tried the new gain (g) option for the voicemail application? From 'show application voicemail': g(#) - Use the specified amount of gain when recording the voicemail message. The units are whole-number decibels (dB). I haven't had a chance to try it yet, but I hope it works. Incoming voicemail messages from analog ports are extremely quiet. Nope: news to me. I'll give it a shot. Thank you! Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording voice messages in mp3 format
[EMAIL PROTECTED] wrote on 11/16/2005 09:46:17 PM: Hi, Yes, I'm using wav for my recording and the file is quite large. I too am using WAV files because of the volume issue: WAV files are shifted two bits louder than any other format. (slight details here: http://lists.digium.com/pipermail/asterisk-users/2005-September/124570.html) That means I'm getting 1MB voicemail files e-mailed to me. It's not that big of a deal, but it's kind of stupid. Any word on if the shifted-bits nature of the WAV format is present (or even better: selectable) on any of the other formats in 1.2? Or might this be a post-1.2 feature worth considering? I haven't programmed in C in a good 10 years or so. But this one motivates me. I'm pretty sure I can hack a simple x = x 2; in there, but making it selectable from within the voicemail configuration file (or wherever else it should go) would be much tougher for me. Might there be a similar selectable feature that I could examine and get an idea of how to do it? Any voicemail experts out there? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modifications to Voicemail
Hello! In honor of 1.2 being released, and now that I'm in the mindset to go spelunking into Asterisk code to address minor annoyances, I have a second issue: Every voicemail system I'm aware of (my Sprint cellphone voicemail, Nortel systems, InterTel telehpone sytsems and others that slip my mind) all use 1 to bypass the voicemail greeting. Asterisk's voicemail uses #. That annoys me. I want to change it. From what I've seen, there is no configuration file for Voicemail that allows you to override the keys that are used within it. Is that true? Are they really hard-coded with the application? If not, how *do* you configure them? If that is so, I would like to start playing with that: first changing the hardcoded value, and then coming up with a way of setting that within the configuration file. I'm no real C programmer, but it motivates me, so I'd at least like to figure out what it would take to do it. Is there any documentation (outside the source, of course) that might guide me a bit? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternative voiceprompts (new subject)
[EMAIL PROTECTED] wrote on 11/15/2005 05:42:44 AM: Olle E. Johansson wrote: be seen as a sample of a full prompt set and something that is extremely This actually leads to a question I've had for a while: Is there a list somewhere of all the prompts (by filename) and what is said? I've searched the Wiki but haven't found anything. Having a list would be a lot easier than transcribing each file. :) http://www.voip-info.org/wiki-Asterisk+sound+files Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hobby box
[EMAIL PROTECTED] wrote on 11/15/2005 02:53:54 PM: Ending last year I used\sold several hundred of product#: FM-INL92SW. Google for it...you'll find some for cheap. Along those lines: are there drivers for the X100/X101 that allow it to act as a normal v.92 modem, even if it's just under Windows? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hobby box
[EMAIL PROTECTED] wrote on 11/15/2005 06:42:40 PM: That being said, and as I mentioned earlier, your cheapest choice is to go to eBay and search for X100P. Here's a question: why are you building your hobby box? To gain practical experience? Then forget the X100: it's like learning Windows NT: yeah, the information might be somewhat valid today, but it's way obselete. The X100 is dead, and very unlamented. If you're just trying to build a fancy answering machine with no other purpose, then fine: waste time on the X100. But if there is any purpose to this, forget the X100P. This coming from the guy with 3 of them on the shelf, collecting dust, purchased for $100 each! However, IMNSHO, your best choice is to shell out somewhere around $100 for a Sipura SPA-3000. This will provide a way for you to connect your home phone line to asterisk, and a way for you to connect an analog phone to asterisk as an extension. You've got three choices, really, on the low end: Digium TDM400 with one FXO port: $130 Sipura SPA-3000 with one FXO and one FXS: $100 Grandstream Handytone 486 with one FXO and one FXS: $80 Yes, you *can* buy an X100P clone for like $10 plus SH on eBay. Don't. You could get just as much experience if you took the $15 you'd spend on the stupid thing and give them to a VoIP provider like Junction Networks or TelIAX and use it. It would be a good idea for you to spend some time on google, voip- info.org, asterisk.org, asteriskdocs.org, etc. searching for information. Seconded. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CentOS vs. Vanilla Kernel
[EMAIL PROTECTED] wrote on 11/07/2005 01:17:31 PM: HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Will you be using Zaptel hardware? The only way I can get zttest results of 100% is with a CentOS 2.4 kernel. Any CentOS 2.6 kernel I've tried (Uni, SMP, with IOAPIC enabled or disabled) gave me 99.99% at best... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timestamps in Console?
[EMAIL PROTECTED] wrote on 11/03/2005 11:49:12 AM: Use 'timestamp=yes' in asterisk.conf instead of -T. This is exactly what I was looking for. Thank you very much! Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sill looking for a provider
[EMAIL PROTECTED] wrote on 11/04/2005 04:34:18 PM: Try calleveryone.com Yes.. I have blown their trumpet before. They are a very good company with great support. Do they support IAX or just SIP? I've been reluctant to use a SIP provider for a number of reasons, including difficulties in using it through a NAT firewall or the fact that I have to open 10,000 ports on my Asterisk server! Am I overly paranoid? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timestamps in Console?
[EMAIL PROTECTED] wrote on 11/03/2005 11:53:17 AM: Chris Wade wrote: Use 'timestamp=yes' in asterisk.conf instead of -T. -T only affects messages generated by THIS connection (ie asterisk -RT generated messages... not server generated messages. 'timestamp=yes' affects all messages generated. And after adding timestamp=yes to asterisk.conf, don't forget to restart asterisk (not just the console) to make the change take effect. Where should it be added? Mine is a default setup. It has two sections: [directories] and [files]. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on TDM - Solved!
[EMAIL PROTECTED] wrote on 11/01/2005 09:05:22 PM: Hello! A couple of weeks ago I mentioned echo issues I was having. It turns out that the echo was only happening for the first 30 seconds or so, so the echo cancellers *were* working, just not training well. I wish they had told me that 2 weeks ago! So, over the weekend, I made two changes: 1) Updated to latest CVS HEAD (from about 3-4 weeks ago), BUT left KB1 as the echo canceller I actually thought I had updated it to MG2, but when I checked zconfig.h, it looks like I didn't! This has now been upgraded to MG2. Users have reported no change: no echo before, no echo now. 2) Changed echotraining=yes to echotraining=800 I think this was the one the fixed it for real. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timestamps in Console?
[EMAIL PROTECTED] wrote on 11/03/2005 03:33:06 PM: Might be worth it to read the stuff in /usr/src/asterisk/doc and in particular the README.asterisk.conf file. Lots of other good stuff in that directory as well. (Not much need to read the source now.) Thank you. I don't mind being told to RTFM, if you can point out the FM I'm supposed to R. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timestamps in Console?
[EMAIL PROTECTED] wrote on 11/03/2005 09:03:44 PM: Might be worth it to read the stuff in /usr/src/asterisk/doc and in particular the README.asterisk.conf file. Lots of other good stuff in that directory as well. (Not much need to read the source now.) Thank you. I don't mind being told to RTFM, if you can point out the FM I'm supposed to R. There is no FM to read. The above reference is to the directory that comes with cvs-head. If you don't use cvs-head, download it anyway and take a look. You misunderstand. I realized you had *already* told me where the information was located. I wasn't *asking* you to point me to the manual, I was *thanking* you for already pointing me to the manual. But thank you again! :) Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suggestions needed...Motherboard incompatibility with Digium Card.
[EMAIL PROTECTED] wrote on 10/31/2005 02:08:24 PM: Does anyone have any Motherboards to recommend us? Any part numbers for Celeron or P4 ? For the record, I've found that kernel version has a lot to do with it, too. CentOS 3.4 gives us 100% zttests. CentOS 4.2 gives us 99.9something% tests. Motherboard: Migrus M4-845Q Mini-ITX. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo on TDM - Solved!
Hello! A couple of weeks ago I mentioned echo issues I was having. It turns out that the echo was only happening for the first 30 seconds or so, so the echo cancellers *were* working, just not training well. I wish they had told me that 2 weeks ago! So, over the weekend, I made two changes: 1) Updated to latest CVS HEAD (from about 3-4 weeks ago), BUT left KB1 as the echo canceller I actually thought I had updated it to MG2, but when I checked zconfig.h, it looks like I didn't! 2) Changed echotraining=yes to echotraining=800 And for the last two days, the users have reported no echo. They are much happier. I thought I'd share. Now, I have a *bad* distortion problem. I will put that in a separate e-mail. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Timestamps in Console?
Hello! Lately, I've been keeping a close eye on an Asterisk box by staying logged into the console for long periods of time. However, it can be very difficult to know how long a telephone call lasts when this is all you see: -- Executing Dial(SIP/SIP105-8e34, Zap/g2/Number|60|t) in new stack -- Called g2/Number -- Zap/5-1 answered SIP/SIP105-8e34 -- Hungup 'Zap/5-1' Did that telephone call last only a few seconds because there was a problem, or a few minutes because there wasn't? It's impossible to tell. Is there a way to add timestamps to each line in the console so you know exactly how long a call took? Or is there another way of telling directly within the console? Thank you very much! Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timestamps in Console?
[EMAIL PROTECTED] wrote on 10/31/2005 08:53:35 AM: On Monday 31 October 2005 09:30, [EMAIL PROTECTED] wrote: Is there a way to add timestamps to each line in the console so you know exactly how long a call took? Or is there another way of telling directly within the console? Of course it's possible, but you'll be maintaining the patch yourself. :-) Why not just enable logging and watch the logfile? You'll get full timestamps there with each line. 'Cause I can't do things like show channels? Yeah, yeah, I could have two windows open, I gues... Me? Lazy? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel + RH3?
[EMAIL PROTECTED] wrote on 10/29/2005 04:01:26 PM: If I add this symbolic link creation into the statup scripts then like I said zaptel working fine, however this is obviously not the right way to fix this issue. Are you doing make config when you compile Zaptel? It does all of this for you... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk using tdm400p has echo
[EMAIL PROTECTED] wrote on 10/27/2005 08:22:04 AM: If you do an fxotune and all of the coefficients are 0, does this mean that fxotune is not making any changes? Based on what Matt has mentioned previously, fxotune only sets the impedence to proper values today. He has not implemented the code to set the coefficients as yet, therefore the expected values are zero's. Are these settings persistent across reboots? The README for fxotune seems to mention that you need to do a fxotune -s in order to reload the card with the analyzed settings (rather than take the 20 minutes it seems to take on my 6 lines). However, if fxotune.conf is all 0's, I sure hope that the settings are persistent on the board! :) Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk using tdm400p has echo
[EMAIL PROTECTED] wrote on 10/27/2005 03:11:11 PM: Are these settings persistent across reboots? The README for fxotune seems to mention that you need to do a fxotune -s in order to reload the card with the analyzed settings (rather than take the 20 minutes it seems to take on my 6 lines). However, if fxotune.conf is all 0's, I sure hope that the settings are persistent on the board! :) The results of fxotune is written to /etc/fxotune.conf; I don't believe they are read back in unless you build something into a bootup script. Correct. From the README: It will write a configuration file to /etc/fxotune.conf. You will need to have your system run fxotune with the -s flag (`fxotune -s`) to set the module with the previously discovered values from fxotune.conf for it to take affect, so essentially if each time you reboot the machine you need to run `fxotune -s`. You might consider putting it in your startup scripts some time after the module loads and before asterisk runs. However, my fxotune.conf contains only 0's for all 8 of each of 6 lines. I'm wondering does that mean that fxotune had no effect, or that whatever effect it does have is A) Persistent within the card between reboots and B) Not reflected by a fxotune.conf filled with 0's... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk using tdm400p has echo
[EMAIL PROTECTED] wrote on 10/26/2005 05:09:30 PM: On Oct 26, 2005, at 3:40 PM, Mark Quitoriano wrote: Hi list, i'm having a problem with asterisk+pstn termination, i just bought a TDM400p and connect my phone line(bellsouth) now when im using the pstn through asterisk there's an echo, i don't know if this is already have been resolved. If it does please point me to the instruction how to resolve this. Try reading README.fxotune and using fxotune to see if it improves it. If you do an fxotune and all of the coefficients are 0, does this mean that fxotune is not making any changes? I've got 6 lines that are coming from a channel bank into two TDM cards and have significant echo, even with Asterisk HEAD and KB1. I just ran fxotune, and all 6 lines came back with all 0's in fxotune.conf... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
How to tell what EC is in place (Was: RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600)
[EMAIL PROTECTED] wrote on 10/24/2005 08:09:27 PM: Also, if you're not already, try using the kb1 echo canceller from CVS-HEAD without aggressive cancellation before taking time to do any of the above. It can be dropped into stable if needed by just copying it (and the contents of the header file) over the top of the mec2 files. Is there a way to tell which echo canceller is in use? I've checked zconfig.h, so I'm pretty sure it's KB1. however, before I experiement with other echo cancellers, I'm hoping there is a way to tell from the Asterisk console which one is active. So I don't lose track! :) Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Definitive answer: time-range includes
Hello! I have a question regarding time-based includes in the dialplan. How are boundary conditions handled? And is there a definitive, documented procedure for how to handle overlapping time includes? For example, if I want to have day/night service from 8 A.M. to 5 P.M., there are two ways I can do it: No overlapping times: [PSTN] include = PSTN-Nighttime|17:00-7:59|mon-fri|*|* include = PSTN-Daytime|8:00-16:59|mon-fri|*|* include = PSTN-Nighttime|*|sat-sun|*|* On this one, no time has more than one valid time entry associated with it. Each entry ends the previous minute to the next one starting. This is the correct one to use iff Asterisk matches the time to the includes inclusive of both the start and end time. However, if Asterisk is *not* inclusive of, say, the end time, you will end up with one-minute holes at the boundaries. Overlapping time: [PSTN] include = PSTN-Nighttime|17:00-8:00|mon-fri|*|* include = PSTN-Daytime|8:00-17:00|mon-fri|*|* include = PSTN-Nighttime|*|sat-sun|*|* On this one, there are two times that have more than one valid time entry associated with it: One minute at each of 8:00 A.M. and 5:00 P.M. Each entry overlaps the other. This is the correct one to use iff Asterisk matches the time to the includes inclusive of the start time but *exclusive* of the end time (or vice-versa) However, if this is not the case, you will end up with one-minute overlaps at the boundaries. The following link in the Wiki describes how to use Asterisk's time-based includes, and even includes overlapping time entries, but does not document these corner-cases: http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+openhours Does anyone know the difinitive anser to how Asterisk matches time ranges? A difinitive, precise statement of what happens with overlapping includes would be nice. Given the right answer, it would allow nice and easy holiday nighttime includes: include = PSTN-Nighttime|*|*|25|dec ; Night on Christmas include = PSTN-Nighttime|*|sat-sun|*|* ; Night on weekends include = PSTN-Nighttime|17:05-7:55|mon-fri|*|*; Night on evenings include = PSTN-Daytime|*|*|*|* ; Default: Handle as daytime Even a simple order-based priority would be great: whichever time entry matches *first* (or last) would work well. In fact, it may even already work that way. I just don't see anything describing how boundaries and overlaps are handled. I'm also not sure that that final entry will work as an include only if there is no other time entry matching. I would hope so. You can't use an include *without* time matching, otherwise it would always be included and would mess things up. Does anyone know the answer to this? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New TDM Revision in the wild: J
Hello! Just thought I'd let everyone know that a new revision has popped out from Digium: Rev J. I don't have an I board in front of me to compare with, so I can't tell you what's different (besides a bunch more text on the back). It looks like there is a PE-68624 chip near each RJ-45 connector now. Google says that it's a frequency control filter The RJ-45 sockets themselves seem different from what I remember as well. All-in-all, the board seems to have a better look to it. Or maybe I'm just tired! :) Does anyone know what might be different about this card? I bought it to put into a new machine that I wanted to be identical to one I just put tougether two weeks ago. Unfortunately, they're not identical now.. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calibrating both RX and TX gain?
[EMAIL PROTECTED] wrote on 10/16/2005 07:49:38 AM: Here's a couple of ways to determine levels... 1. using the model 4 transmission test set, attach the tone generator to one analog pstn line and the transmission level test jacks to a second pstn line. Dial from one line to other and measure the tone. Divide by two, and the result is the loss associated with a single analog pstn line from your location to your central office. Remember, I'm not working with simple POTS lines. I've got an Adtran TA 612 providing CO lines from a T1. There is nothing that says that the RX and TX settings on the Adtran are the same... Therefore, just dividing by 2 won't work. Also, couldn't there be an issue on standard POTS lines where the effect upon a singnal between TX and RX is different? It seems you're just exchanging one set of assumptions for another. But you're the expert! :) 2. use one of those analog pstn lines to dial the distant milliwatt generator (regardless of where its located), and measure the level of the tone. Subtract the loss determined from step #1 and now you have the loss associated with facilities interconnecting your central office all the way to the distant milliwatt generator. This doesn't address the problem above, correct? The end result will be whatever loss values you measure/calculate, you'll still have to play around with the rxgain txgain to minimize the echo while also maximizing the audio levels. The process will become a _qualitative_ eval process, not a quantitative one. It doesn't make any real difference which tools you use to get there or exactly where the milliwatt generator happens to reside. So how important or valuable will getting a milliwatt test number be? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calibrating both RX and TX gain?
[EMAIL PROTECTED] wrote on 10/17/2005 12:45:13 PM: Here's a couple of ways to determine levels... 1. using the model 4 transmission test set, attach the tone generator to one analog pstn line and the transmission level test jacks to a second pstn line. Dial from one line to other and measure the tone. Divide by two, and the result is the loss associated with a single analog pstn line from your location to your central office. Remember, I'm not working with simple POTS lines. I've got an Adtran TA 612 providing CO lines from a T1. There is nothing that says that the RX and TX settings on the Adtran are the same... Therefore, just dividing by 2 won't work. Obviously I _assumed_ you were working with analog pots lines. Sorry. Since I don't have access to your previous/original postings, now I'm somewhat confused as to exactly how the T1 and 612 are interconnected wtih asterisk. Is the T1 terminated on asterisk or the CO? Are the ports on the 612 FXS (for phones) or FXO (for CO lines)? It's a Smart T1: Internet and CO lines on the same T1, which are broken out by the Adtran. We have 6 CO lines: PSTN T1 - Adtran 612 FXS - TDM400 with FXO Modules - FXS modules Ethernet or | local snom 190's V Firewall (to rest of network) My original e-mail, with a lot more detail regarding my problem (way low sound and much echo) is included at the end. An additional point: When I call on a cell phone, there is no echo. Their echo cancellers kill it. Their cancellers are so good, though, that when I use the echo test, all I hear is a very small amount of quiet garbled noise at the beginning of each word. Very impressive! When will Asterisk's echo cancellers get that good? :) Unfortunately, I did not realize that when I installed the system, and I used calls to my cell phone to determine connection quality. Did I mention that the system is about 800 miles away from me now? :( Also, couldn't there be an issue on standard POTS lines where the effect upon a singnal between TX and RX is different? I think I need a better understanding of how your assets are interconnected before I utter more inaccurate statements. From a telco perspective, a customer line (whether an analog pstn copper pair, or T1-extended) should never have a different tx vs rx gain/loss at the rj11 point. Should be exactly the same in both directions. I guess that's kind of the definition of a hybrid? :) So how important or valuable will getting a milliwatt test number be? Fairly important if you want to identify audio quality/level issues. Not so important if you were just trying to adjust rxgain/txgain on a digium TDM analog card. Well, I've got +15db rxgain and -3db txgain. This gives me barely acceptable levels both ways, yet I still have lots of echo. Yet when I put an analog handset on the line, both RX and TX levels are fine. In other words, even if you leave out the large echo I'm getting, why don't my TDM interfaces give me audio levels anywhere *near* what a $10 analog handset gives me? Line loss isn't an issue: there's 12 feet of Cat5 between the channel bank and the TDM card! :) It sure feels like something more than simple levels and delay: something like badly matched impedance. I can't figure out why a handset would sound fine in both directions, when my rx and tx gains have to be *so* out of whack. In any case, you can still use a distant milliwatt generator to obtain realistic measurements, regardless of how you use those measurements. OK, then, with that said: Anyone want to give me a milliwatt test number? The closer to Camden, South Carolina or Detroit, Michigan, the better? :) Thank you *everyone* for all of your help and suggestions. I greatly appreciate any information you can add. Tim Massey Original E-mail: Hello! I'm having an echo problem with a TDM card. The TDM card is being fed by a channel bank just 12 or so feet away. When you put an analog handset on the line, both the RX and TX volume seem to be just fine. However, when I use the TDM card, I have to have an rxgain of 13.5, and even then, the audio is relatively quiet. I'm also getting echo on these lines, so I have turned the txgain down as low as I can and still be heard. Right now, it's at -6, but it will have to come up some because that is too quiet. But I still have echo. I am in the middle of trying to get a milliwatt test line to calibrate the rxgain properly. However, this won't help me with the txgain, will it? How can I properly calibrate the txgain? By ear? Or is there a more scientific method? For example, once I have the rxgain calibrated for all of the lines, could I then call into, say, Zap/3 from Zap/4 and run Milliwatt() on Zap/3 and use ztmonitor on Zap/4 to calibrate it? I'm sure it's not perfect, but would it be close enough? A second question: doesn't it seem wrong that my rxgain and txgain are
Re: [Asterisk-Users] Calibrating both RX and TX gain?
[EMAIL PROTECTED] wrote on 10/12/2005 01:23:57 PM: On Wed, Oct 12, 2005 at 12:05:32PM -0400, [EMAIL PROTECTED] wrote: I am in the middle of trying to get a milliwatt test line to calibrate the rxgain properly. However, this won't help me with the txgain, will it? How can I properly calibrate the txgain? By ear? Or is there a more scientific method? I contacted Rhino to see if they had any suggestions, and they were able to give me a few. What finally worked was setting the Asterisk gains back to 0 for all channels, then adjusting the gains down on the channel banks themselves for the phone (FXS) interfaces only. A huge improvement! My current adjustements are the following: According to the company that installed the channel bank, there is a 0db and -10db setting on the smart jack for the T1. They claim that this was most likely set to -10db by the ILEC when the T1 was installed, and that would be causing the low audio volume. Does this make sense to anyone? Wouldn't the -10db affect the *digital* levels, not the analog waveform encoded within the digital signal? I'm still trying to get a milliwatt test line to calibrate from. They claim that they won't give that out to end users because it could fry the T1 card. Sigh. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calibrating both RX and TX gain?
Hello! I'm having an echo problem with a TDM card. The TDM card is being fed by a channel bank just 12 or so feet away. When you put an analog handset on the line, both the RX and TX volume seem to be just fine. However, when I use the TDM card, I have to have an rxgain of 13.5, and even then, the audio is relatively quiet. I'm also getting echo on these lines, so I have turned the txgain down as low as I can and still be heard. Right now, it's at -6, but it will have to come up some because that is too quiet. But I still have echo. I am in the middle of trying to get a milliwatt test line to calibrate the rxgain properly. However, this won't help me with the txgain, will it? How can I properly calibrate the txgain? By ear? Or is there a more scientific method? For example, once I have the rxgain calibrated for all of the lines, could I then call into, say, Zap/3 from Zap/4 and run Milliwatt() on Zap/3 and use ztmonitor on Zap/4 to calibrate it? I'm sure it's not perfect, but would it be close enough? A second question: doesn't it seem wrong that my rxgain and txgain are so far off when I'm just talking to a channel bank 12 feet away? I sure don't have cable loss. It sure seems like the impedance is way off or something. Is there a way to test this further, rather than just cranking up the gain? My guess is that using the milliwatt line will just tell me to make the rxgain higher, which will probably just make the echo issues worse... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: RHEL / CentOS Enable APIC
[EMAIL PROTECTED] wrote on 10/01/2005 10:17:47 AM: Is there a way with either RHEL or CentOS to force it to use an APIC-enabled kernel? I've tried Googling but no success. I can find no way of doing this during the install. If you have a single processor system, AFAIK you are stuck with standard PIC (not APIC) support. And while APIC and SMP have little to do with each other any more, it seems that only SMP kernels have APIC support. Therefore, you must install the SMP kernel. And again, I can find no way of forcing the install to install an SMP kernel on a uni machine. So, after the install takes place, you must do the following: Mount CD #2 change into the RedHat/RPMS directory rpm -ivh whatever the kernel-smp.whatever.rpm is This will install the SMP kernel and add an entry into Grub. If you wish to boot this kernel by default, modify the /boot/grub/menu.lst and change DEFAULT=1 to DEFAULT=0. Then, reboot. At this point, zaptel will not load anymore: it will complain that it cannot find the module. You will have to recompile zaptel. After this, the zaptel module will load. This works, and it doesn't seem to be too cumbersome, but it sure seems like there should be some sort of installation parameter you could add somewhere to force-load an SMP kernel even on a uniprocessor machine. Of course, even better would probably be compiling a uniprocessor kernel with APIC support, but whatever. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zttool improvement: histogram
Hello! A small suggestion for an improvement to zttest: some sort of histogram to show a broader range of the results that are being returned. For example, on a test machine I ran each of the following items in separate infinite loops at the same time: ssh-keygen -b 8192 -t rsa -f /test.key dd if=/dev/zero of=/test.file bs=1024k count=5000 while in a third console I ran zttest. I did this twice each over several hours. My results were encouraging: Best case was 100% (and from watching the output from time to time there were lots of those), average was 99.90% for one and 99.98% for the other, but the worst-case was troubling: 83% for one, and 68% for the other! Of course, over several hours, there were tens of thousands of results, and even a single bad result will throw off the worst-case result. Hence, the request for some sort of histogram: something that would show how *many* results were way off, and by how far. Something that would show the nature of the bell curve I would expect to get. Of course, I could probably parse the raw output of the zttest command with something to plot this. However, my unix-fu is not good enough to do that. Does anyone have a suggestion? Or would this be valuable to have in the zttest command internally? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] kjournald and zttest results
Hello! While performing some zttest's for some time today, I was also keeping an eye of a top of the machine. While the zttest was running, I also had a ssh-keygen and a dd creating a 5GB file on an EXT3 partition running. I noticed that for the most part, I got a decent number of 100%'s, and a bunch of 99.6%'s or higher. However, it seems that whenever the zttest dropped below 99%, it was usually (or at least often) because kjournald was jumping up in the top list. Has anyone done a comparison of zttest results with journalled and non-journalled file systems? From my very limited testing today, it seems like that might be an area to investigate. Does anyone have any suggestions on how one might best test this? Or adjustments that might be made to journalling settings that might improve this? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: RHEL / CentOS Enable APIC
Hello! I'm setting up Asterisk on a new system. In the past, all of my Asterisk boxes have either been embedded-style systems that do not supoort APIC, or multi-processor systems where APIC comes along with SMP. However, now I'm trying to install Asterisk on a single CPU (and non-HT) system that does support APIC (A P4 Northwood an Intel 845 chipset). I've used both RHEL3 and RHEL4 (and CentOS 3 as part of [EMAIL PROTECTED]). For the life of me, though, I cannot seem to get an APIC enabled kernel installed. It seems that because it's a uniprocessor system, the default is to load a uniprocessor, non-APIC kernel. On the 2.6 kernels I've tried adding lapic as a kernel parameter, but it does not help. I must say that I'm surprised that [EMAIL PROTECTED] doesn't do this automatically, given the benefits of an APIC-enabled kernel for Asterisk. Is there a way with either RHEL or CentOS to force it to use an APIC-enabled kernel? I've tried Googling but no success. Thank you very much for your help! Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?
[EMAIL PROTECTED] wrote on 09/30/2005 01:10:34 PM: On Fri, Sep 30, 2005 at 01:32:07PM +0100, Angus Comber wrote: Hello I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this likely to be enough power for a 8 extension system with 6 external pstn lines? Probably, yes. I use a 533MHz to handle two outside lines and 4 internal extensions with no problems. There is very little CPU usage even with nearly everything in use. However, I'm doing no transcoding and no-compression codecs. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Problem setting up TDM22B card
[EMAIL PROTECTED] wrote on 09/27/2005 03:13:21 AM: Hi, I did dmesg | tail it says ... dmesg | tail f6 != 58 f7 != 59 f8 != 58 f9 != 59 fa != 58 fb != 59 fc != 58 fd != 59 fe != 58 Freshmaker failed register test The only time I've seen this it has been on a PCI 2.1 computer. On a PCI 2.2 computer, I did not see this. It also was a early TDM board. If you have a pre-Rev F board, you may want to swap it for a newer one. I am pretty sure that this error was fixed by moving from an earlier board to a Rev F. I have a Rev H now, with no issues. I have not been following this thread closely. Which chipset does your motherboard use? For the record, none of the desktop or server Intel 440-series support PCI 2.2. (Technically, a single mobile chipset, the 440MX, does support PCI 2.2) However, all of the 800-series chipsets do. The easy way to figure this out for Intel chipsets is: 1) Does the motherboard use slot processors? If so, it's PCI 2.1. 2) Does the motherboard support 133MHz PIII processors? If so, you're possibly PCI 2.2. 3) Pentium 4 chipsets are all PCI 2.2. I have no idea what other non-Intel chipsets support PCI 2.2. Reference: http://www.intel.com/design/chipsets/mature/450_440.htm http://www.intel.com/design/chipsets/mature/index.htm Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [MSG]TDM Error on ASUS Pundit-R
[EMAIL PROTECTED] wrote on 09/27/2005 01:18:35 PM: Hi I have looked around but I cant find an answer for this, I randomly get the error 'TDM PCI Master abort' and the system locks up. All I have found so far are a couple other posts on it but no solution. Running fedora core 3, asterisk stable, zaptel stable. I had this problem with a Rev F board. Upgrading to a newer board fixed it for me. I don't know if anyone has a more specific solution... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM card and voicemail volume
[EMAIL PROTECTED] wrote on 06/28/2005 07:52:50 AM: I was able to raise the volume from inaudible to acceptable by increasing the RxGain in zapata.conf by 5db. I'd rather not go the uncomressed wav route, as it will chew up storage in my email system. I know I'm way behind on reading this, but I thought I would follow up. According to this message: http://lists.digium.com/pipermail/asterisk-users/2004-November/072990.html the reason that uncompressed WAV files are louder is that the software that saves the WAV file is amplifying the volume of the files by shifting the data two bits to the left (or making it 4x louder). It is in no way fixing the underlying problem of the file being too quiet; it is just throwing away dynamic range in order to amplify the file. Now that may not be a bad solution: if you don't need the dynamic range, but you *do* need the volume, so be it: you would prefer the off-chance of some clipping. It *has* to be a better solution to using the rxgain setting if you don't need to: rxgain is going to affect echo for the worse. Also notice that the volume of these files is sufficient when they are played back over the telephone: it's only when you play them back via a sound card that you have the volume problem. So, you can't just willy-nilly amplify everything. Hope this helps. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B + Voicemail poor Quality
[EMAIL PROTECTED] wrote on 07/18/2005 11:56:06 AM: Recently, I installed TDM04B 4 FXO card on to my Asterisk box and installation went perfect. The only problem I am facing is the Voice mail has very poor quality when any users leave voice message via PSTN line. We can not hear either from the extension nor from the WAV email attached. Has anyone experienced this problem before, please help? Yes, its well known. See bug #2023 in the bug tracker. Kevin would like to address this along with addressing missed frames (common complaint when attempting to use spandsp) before the next formal Stable release. But, its too early to guess whether that will actually happen (as of right now). Does this relate to the e-mail you just sent to the list regarding volume issues with voicemail files not saved as uncompressed WAV's? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Differences between test equipment
Hello! Given the current discusison regarding ztmonitor, line testing, etc., I've been looking into purchasing a used transmission test set. From my research, it seems that there are two items that might fit the bill: the HP 3551A and the HP 4935A. I know nothing about these specific devices. I *do* have a good background in electronics, and I understand the concept what they're measuring and why., but I know nothing about the specifics of how this relates to transmission test sets! :) In fact, I'm not even sure that these are indeed the right devices for the job. Could someone who is familiar with either of these devices tell me if they will fit the bill? And if possible, which of these has more useful features as telephone line test equipment? I'm handy with an oscilloscope, function generator, VOM, etc. Will I be able to drive either of these? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???
[EMAIL PROTECTED] wrote on 04/19/2005 01:32:57 AM: ** Extract begins ** SCSI RAID can cause the problem. If disabling hyper threading does not resolve your problem my next suggest would be to revert to a PATA IDE hard drive solution configured to UDMA level 2 using hdparm. SCSI or SATA causes problems on some systems from what I have seen. The problem increases when using a SCSI or SATA RAID. ** Extract ends ** I really hope that they are wrong, as I don't feel like throwing away my nice expensive Ultra320 SCSI RAID controller and hot plug drives and replacing them with some crusty old IDE config. Needless to say I'm not going to go and shell out on IDE controller drives until I'm a little more certain that this is actually a problem and have asked them for more information. Does anyone else find it odd that the TDM could possibly have a problem sharing a box (but not an IRQ) with a SCSI controller? Yes. It has to do with latency and bus contention. I've run a TDM board in an IBM Netfinity 5600 server with an IBM ServeRAID 3L controller (SCSI-U2W). The big difference, though, is that the RAID controller was on its own PCI bus, and the TDM card was on its own PCI bus. With both controllers on the bus, you can have latency issues. For example, if the RAID controller sets up a DMA of a big chunk of disk, it owns the bus for that transfer. If an Ethernet packet is delayed by 50us during that time, nobody cares. But if the TDM card is delayed, it most certainly cares: especially as its generating 1000 interrupts a second! That's the problem with the TDM cards. They do *nothing* on the CPU side. The CPU has to do *everything*, and it has to do it *immediately*. When you are using plain-jane IDE, you can tweak the kernel to put the IDE stuff at a low priority. But when you've got a fancy RAID controller, it tends to think it's the most important thing in the system. And as a rule, hard drive I/O usually *is* the most important I/O going on in a system. However, in this case, the TDM card trumps that. And Digium doesn't know how to tweak every last RAID driver in existence for low-priority operation--or even if it's possible. Hence, the recommendation for IDE. Combined with the fact that they have also recommended that we turn off hyper threading (also causes problems with TDM, apparently), I'm wondering if these cards shouldn't come with a warning not to use anything with half decent performance in your * server! Yet they require PCI 2.2, which eliminates most Pentium III's and lower! :) I'm still in the midst of testing the TDM cards. So far, so good, in an EPIA-based solution and in the 5600. But I've been through at least half a dozen different systems before I've found these... Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Illegal instruction (core dumped)
[EMAIL PROTECTED] wrote on 04/17/2005 11:02:51 AM: On April 17, 2005 05:55 am, Tom Fanning wrote: Illegal instruction (core dumped) Sounds like you have compiled asterisk for a processor that is greater than the processor you're running on. I.e. compiled and told it to use P4 instructions when you're on a P3, or maybe even told it to use MMX on a Via processor... This is especially true for Via processors. They identify themselves as 686 processors, but do not implement the CMOV instruction, which GCC considers to be a 686-class instruction. Do a search for Via CMOV Linux compile or somesuch on Google and you will see the modifications you will need to make to the makefile to address this. Incidentally, I believe that the latest processors (the Nehemiah C5P found on EPIA MII boards) support CMOV. I'm less sure, but the Nehemiah processors themselves may also support CMOV. The Samuel processors, though, do *not* support CMOV. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone had problems with Digium TDM400P and hyperthreading?
[EMAIL PROTECTED] wrote on 04/15/2005 01:45:22 AM: Digium have told us that a problem that we are having (with accuracy of zap interface as measured using zttest) may be due to the fact that we have a Xeon processor with hyperthreading and have suggested turning H/T off. I've never ran Asterisk on an HT-enabled processor. However, I've had too many problems to count with HT and Linux. I turn it off on nearly every server that has it. Then again, most of my servers are not CPU bound and I couldn't care less about the performance. Also, make sure you update your motherboard's BIOS. It's responsible for updating the CPU microcode, and often the BIOS may have newer microcode than your Linux distribution. Anyone else experienced a problem like this? No too keen about turning H/T off, as we're running the SMP RH kernel and don't really feel like replacing the kernel (and other kernel-specific bits) on the off chance that H/T is actually the problem. An SMP kernel should run just fine on a single-processor box. Slower, but fine. At least, it works for me... TIm Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card
[EMAIL PROTECTED] wrote on 04/12/2005 11:36:47 PM: [EMAIL PROTECTED] wrote: In other words, a PCI-based co-processor would double the PCI bus bandwidth necessary. And with a latency-sensitive product like voice, bus contention is not something you want to add to! :) It only 'doubles the bandwidth required' when compared to a single-board solution, which does not exist. My statement was not meant as a criticism: only a description as to the difference beween putting the coprocessor on the DS3 board versus putting it on the PCI bus. As someone who has no need for a DS3 board, I am not familiar with whether there is a card that does everything on a single board. I was just describing the difference in response to a question. When compared to doing the transcoding and echo can in the host CPU, it would be a major win :-) Ah, the magic of DSP's! :) There's no question that you would be challenged to do a DS3-worth of transcoding and echo cancelling with a general-purpose CPU (or even several). Also, keep in mind that a DS3 is _only_ 45 megabits per second. Any PCI bus (even lowly 33MHz 32-bit PCI) can easily handle 90 megabits per second of traffic. People looking a DS3 cards are also likely to deploy them in servers with multiple independent PCI buses, which would then allow for even more bandwidth. There is no question about this. Base PCI can handle a theoretical maximum of 132MB (That's *bytes*) per second. A DS3 with separate co-processor board is a tiny part of that: about equivilent to that of a 100Mbit Ethernet controller. Old hat. The only issue is latency. Either you transfer information in big chunks efficiently, or small pieces inefficiently. Given that there will be at least three devices participating on the bus (the CPU, the DS3 card and that theoretical co-processor), that means bus contention. If you're talking 33MHz 32-bit PCI (which, from the picture, seems to be what we're talking about here), you may run into problems when you add in the Ethernet controller, disk controller, etc. Of course, high-end hardware makes this less of an issue: if you can dedicate a PCI bus to the two cards, then go crazy! And if you actually *need* to manage 672 channels, you can afford a decent server with dual PCI busses! :) The mind boggles at the possibilities! My mind boggles at the need for a DS3 in the first place. I thought I was pretty cool the day I got my first T1! :) Some of us have to slum it for a living... :) Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card
[EMAIL PROTECTED] wrote on 04/12/2005 10:51:49 AM: Andrew Kohlsmith wrote: secondary card for DSP functions is very inefficient of the PCI bus. I'd be curious to know if the Digium cards can even do PCI-PCI DMA. The Digium TDM cards can DMA into any RAM accessible over the PCI bus, regardless of whether it is located on the motherboard or on a PCI card. That's not the point. The point is that you have to transfer voice data twice: once from the DS3 card to the co-processor, and once from there to the eventual destination (probably system RAM). If the co-processor is integrated into the DS3 card that first transfer is handled and echo-cancelling is performed *before* the data hits the PCI bus. In other words, a PCI-based co-processor would double the PCI bus bandwidth necessary. And with a latency-sensitive product like voice, bus contention is not something you want to add to! :) Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP to the PBX
[EMAIL PROTECTED] wrote on 04/01/2005 12:36:07 AM: I'm new to the VOIP world and need some advice. I currently have a premium/ full functioned Panasonic PBX installed in my house/ small office... and have some extra unused telco lines available on the PBX. I'd like to use one of these extra lines for VOIP into the PBX/ phone arrangement. Can I set up Asterisk to do this? I have a spare computer and a Digium wildcard x100p card. You would need an FXS interface (the TDM400), not an FXO. The Panasonic has an FXO interface, just like the X100P: they're both designed to plug into PSTN lines. You need something that *generates* a PSTN. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID on voicemail messages
[EMAIL PROTECTED] wrote on 04/01/2005 09:04:38 AM: Take a look at the voicemail.conf.sample that comes with asterisk. Inside you will see how to change the voicemail email message that is cerated and add the phone number (and remove the name) for callerid. Thanks. Once I found that it was the name portion of CallerID, it made it easier to find the solution. At first, I couldn't figure out where the Toll-Free Caller was coming from... Sorry for the RTFM question... Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this possible?
Paul wrote: I'd like to setup my Asterisk box to receive a call on the incoming POTS line and immediately redirect back out to connect to another phone number. Im thinking I could use either the threeway feature of that POTS line, or a second POTS connected to a different FXO card. Does ANYONE know if this is possible and if so, how it's accomplished? Three way calling would be interesting (and maybe impossible), but doing that with two POTS lines (or a POTS line and a VoIP provider, or just a VoIP provider, even) is trivial with Asterisk. You would accept the call from line #1 and dial out via line #2 to whoever. When the remote end picks up, the calls are bridged. Asterisk does this all day long. In fact, it's really one of the only two things Asterisk does (the other being play audio for and receive audio from a channel). It's incredible the amazing things you can do with a system that, in the end, really only does those two things! Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Issues with ringing on FXS ports
[EMAIL PROTECTED] wrote on 04/01/2005 03:24:49 PM: Try adding the module parameter boostringer=1 when loading the wctdm driver. This raises the ringing volts to 89V peak. Is there a list of these anywhere? This is now the third one I've heard of, with no documentation: lowpower (IIRC), robust and now boostringer. Do I have to go diving in the source, or is there a Wiki I can't find? Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
[EMAIL PROTECTED] wrote on 03/31/2005 02:24:11 PM: -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Its an odd thing. Some people have to reload, others don't, and there has been no effort to determine why it occurs. I've got two systems that do have to be reloaded regularly. Go figure. These kinds of erratic interoperability problems often speak to a marginal design. If you're a little short on filtering, or your signal levels aren't quite right, or something like that, it's easy to end up in a situation where your product will work great under optimal conditions, but fail erratically out in the field. Oddly enough, I read this as I was working on adding an X100P to a computer. Every time I jiggle the cable, I get a slew of FXO PCI Master abort errors! Kind of epitomizes that exact idea! :) Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID on voicemail messages
Hello! When someone calls into my toll-free number delivered via IAX, the caller's number shows up on my SIP phone. However, when I receive an e-mail voicemail message, I get this message: Just wanted to let you know you were just left a 0:01 long message (number 2) in mailbox 200 from Toll-Free Call at ... Why doesn't the Caller ID show up properly in the voicemail when it *does* show up on my phone? I don't even know where to begin to look for this one! :) Thank you for any assistance you might be able to provide. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID on voicemail messages
[EMAIL PROTECTED] wrote on 04/01/2005 12:03:15 AM: How do you get it to say where its from in the first place? ;-) It just does! :) I've never done anything to enable it: It just happens automatically. For clarification, this is in an e-mail sent to me when I receive a voicemail. This is not in a voice prompt or anything like that. From further research, it seems that the issue is that my phone is using plain Caller ID (just numbers) and the Voicemail app is using the name portion of Caller ID+Name. Unfortunately, my provider is not providing the name portion: they're just sticking Toll Free in there. Is there a way to tell the Voicemail app to just use the number portion? Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Physically Small Box Asterisk Systems
[EMAIL PROTECTED] wrote on 03/30/2005 01:46:10 PM: Looking for reccomendations for a physically small box configuration that will do: Run Asterisk One T1 Card One LAN port Enough CPU power to handle encoding/decoding all 24 T1 channels to/from G.729a Someone mentioned the mini-ITX systems, but there seemed to be a concern about adequate CPU power for doing transcoding of more than a few channels. There are P4-based Mini-ITX boards that should handle that just fine: http://www.caseoutlet.com/shopexd.asp?id=207 I *love* the Mini-ITX format. The VIA CPU leaves a lot to be desired performance-wise, but the format's nice. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which analog phones to use and why?
Hello! Now that I finally have my TDM board working, I want to move forward with using PBX functions. However, it seems cumbersome to use standard POTS telephones with Asterisk. I know that there are many of you installing even large systems based on channel banks and analog telephones. What phones are you using? How do you simulate phone system features on a phone that doesn't have extra buttons? Or are you all using ADSI telephones? It seems that for the price of a ADSI telephone (never mind the cost per channel of a channel bank and T1 card), you can get a good quality IP telephone. In that case, what is the appeal of analog? Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which analog phones to use and why?
Steven Critchfield [EMAIL PROTECTED] wrote on 03/28/2005 11:44:03 AM: Depends on what functions you are trying to implement. Hold isn't hard on a regular phone. Transfer isn't hard. Voicemail access isn't hard. Beyond that, there isn't a lot that needs to be done. If you find that you need more functions, then you may need to move up to a SIP phone. Well, what it seems to come down to is two things: 1) People *expect* business phones to just plain have more buttons 2) People want one-button convenience For example, people want to be able to push a single button to reach at least a selection of internal extensions. Or, they want to be able to press a single button for parking a call, or voicemail, or who-knows-what. Of course, a standard analog phone can't do those things: it doesn't have the buttons! :) I guess even a telephone with speed dial buttons could do that, maybe? Something like this: http://www.101phones.com/flypage/2126/8a3a9cb7ed9a26e52f4129070e30b829/Panasonic_KX-TS105W I was just wondering how others are addressing this. You can't all be making receptionists memorize codes, are you? :) Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with 1 dialing (recording says must dial 1 when I thought I did)
[EMAIL PROTECTED] wrote on 03/28/2005 01:19:03 PM:
TRUNKMSD1=1 ; MSD digits to strip
(usually 1 or 0)
TRUNKMSD2=2 ; MSD digits to strip
(usually 1 or 0)
; logn distance calls
exten = _91NXXNXX,1,NoOp(Dialing: ${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,3,Congestion
Your dial command is stripping the one. That's what the ${EXTEN:1} part
does. So, yes, you are dialing the 1, but the dial command is stripping
it.
If you want to keep the one, use this:
exten = _91NXXNXX,2,Dial(${TRUNK}/${EXTEN})
When I dial a long distance number (916503270309 for example) I get the
message (I think from SBC) saying I must first dial a 1. Other times,
it works, like when I dial this number (914082341389).
I have no idea where you're located. Is it maybe that you have 10-digit
dialing and that the one that works is a local call, and therefore does
not need the 1?
Tim Massey
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Re: [Asterisk-Users] problem with 1 dialing (recording says must dial 1 when I thought I did)
[EMAIL PROTECTED] wrote on 03/28/2005 03:24:50 PM:
[EMAIL PROTECTED] wrote on 03/28/2005 01:19:03 PM:
; logn distance calls
exten = _91NXXNXX,1,NoOp(Dialing:
${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,3,Congestion
Your dial command is stripping the one. That's what the ${EXTEN:1} part
does. So, yes, you are dialing the 1, but the dial command is stripping
it.
No, your command is correct: you need to strip the 9. Sorry about that.
Time for more coffee! :)
Tim Massey
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Re: [Asterisk-Users] call center: agents, queues, sip
[EMAIL PROTECTED] wrote on 03/28/2005 11:50:25 PM: Why does the agent has to be always connect? Is there a way to close the connection and have * to call the correct agent when a call arrives? If you want this to work through NAT, the soft clients will have to keep a connection open. That's the only way to keep a NAT tunnel open... If it's *not* going through NAT, you may not need the connection; however, if you're worried about the volume of connections overloading a box (and the limit is at *least* 32,000), you're probably going to have to deal with NAT. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware question
[EMAIL PROTECTED] wrote on 03/25/2005 09:14:42 AM: Hello I want to to know if the motherboards VIA are fully supporte by asterisk. This is a complex question. The *software* is fully supported. Depending on the CPU you use, you may have to modify the makefiles (some VIA CPU's do not implement the CMOV instruction), but with that change the software will work just fine. However, Digium *hardware* is a different story. The TDM and X100P boards require that the card be placed on its own interrupt. Interrupts are scarce on a VIA platform: there's no IO-APIC, and there's a lot of integrated hardware. It is doable, however. I'm using a TDM board with a VIA EPIA-MII board with zero problems. No clicks, no static, nothing. I'm even sharing an interrupt (the TDM board and an unused (and AFAICT not-diableable) Cardbus controller), and still no problems. However, YMMV... And also, some of those motherboars say that with 1 pci slot , using a special riser card you can connect 2 pci cards. Will that work to have 2 pci cards (FXS or FXO ) on asterisk? Again, a complex question. The short answer is yes, the dual riser in and of itself will not cause a problem. The long answer is that it is highly unlikely that you'll find an interrupt for it. I have the dual riser and the second port wants to use an interrupt that already has a couple of devices on it, including the Ethernet interface. So, that's probably not possible on my system. The really annoying part is that my system has *SIX* unused interrupts: 3,4,6,10,11 and 13. Now I know that two of those are traditionally used by legacy devices (math coprocessor and floppy controller), but what about 3,4,10 and 11?!? I can find no way to get the computer to use those IRQ's. Everything's onboard, so changing PCI slots is not possible. It's frustrating. 15 interrupts is not exactly a lot, but when you ignore nearly half of them, it's real hard to use your motherboard... Tim Massey thank you Fabian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware question
[EMAIL PROTECTED] wrote on 03/25/2005 09:14:42 AM: Hello I want to to know if the motherboards VIA are fully supporte by asterisk. This is a complex question. The *software* runs on Mini-ITX (what I assume you're asking about) just fine. The *hardware* *may* have issues however. These devices do not support IO-APIC, so you can have interrupt issues with the X100P and TDM400 devices. I am running a TDM400 on a Via EPIA-MII board, so far, without problems. No static, no clicks, no buzzing, no erros, nothing. So far... And also, some of those motherboars say that with 1 pci slot , using a special riser card you can connect 2 pci cards. Will that work to have 2 pci cards (FXS or FXO ) on asterisk? Again, a complex question. The short answer is yes. The PCI riser cards will work just fine in and of themselves. However, the odds of you being able to get two interrupts completely free and clear for the use of two TDM boards is slim. This whole IRQ routing issue is a drag. On my system, there are three interrupts completely unused (3, 4, 6, 10, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Toll-free DID switchover: Get status?
Hello! I am in the middle of having a vanity toll-free DID set up. It's been 13 days now (9 business days). This is the first time I'm doing this, and I'm not sure of the process. There has been a very weird progression of changes on my number, from fast-busy, to a message saying that I'm calling from a phone with restrictions (no matter *what* line I call from), to a number advertising a $4/min national directory assistance, and now back to the restrictions error message. Is there a way to track the status of a toll-free DID switchover? I've checked the ATT database and they no longer show my number as being available (and I hope that's because I'm getting it! :) ), but that's all I've been able to find out. The company performing the switchover has been less than proactive in giving me information on this, even after several requests. I understand that much (if not all) of the switchover is out of their hands; however, I would really like to know why my DID seems to be taking such a strange (and seemingly slow) roller-coaster road, and more important, when I'll be able to get off! Thank you for any information you would be able to provide! It seems that toll-free is its own little world, and they don't want others to be involved. I would appreciate any help you would be able to provide. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forklift a 2000 phone PBX
[EMAIL PROTECTED] wrote on 03/24/2005 06:42:24 PM: Does anyone know how to qualify existing Cat3 wiring for use as a LAN? That's easy: Cat3 is able to handle 10Mbit. So if the wire truly is Cat3, you can use 10Mbit switches and be in good shape. Now, how do you know if the wiring is truly Cat3? Just because the raw wire is Cat3 means nothing if they wrapped it around a few fluroescent lights... ;) Your best bet would be to certify the wiring. A used scanner you bought on eBay would be fine: there are plenty of Cat5 scanners around that people are replacing with Cat6 scanners. I like the old Pentascanners (used to be Microtest, now owned by Fluke). They will also certify for Cat 3. The problem is, most phone wire is: A) Terminated into 66 blocks, B) Not ran with data requirements in mind, and C) Often terminated as two lines per wire. For A, you have to re-terminate all of the lines, for B, you may have to re-run some (or even most) of the lines because of quality or length issues, and for C you may have to run fully half of the lines again because they may want two jacks in an office like there is now, but there's only one wire going to that office. You could use those mini-switch-in-a-jack thingies, but they are usually more expensive than it would cost to run more wire! :) In short, unless the phone wire is just a few years old at the *oldest*, assume the worst: the wire will not work out for you. That, by the way, is why *all* wiring I have done for my clients is all done Cat5 (or higher) into patch panels. I then use a patch panel wired to a 66 or (usually a) 110 block for connection to the phone system and plug into it like you would an Ethernet hub. That way, when they are ready for VoIP (or just want to use a data jack for phone or vice-versa), it is idiot simple. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Success report with TDM400 and Via EPIA-MII motherboard
Hello! In spite of a number of complaints, I have tried to use a TDM400 on a Via EPIA-MII motherboard with a 1.2GHz C3 CPU. cpuinfo and interrupts are included at the end of this e-mail. I have had no problems with it so far that I can attribute to the computer. I have, though, had continual problems with a certain telephone... Anyway, this module currently has 3 x FXS modules and 1 x FXO. I have not tested the FXO functionality yet (beyond cursory I-can-ring-out and I-can-receive testing). I have spent several hours calling back and forth between the FXS modules, though, and so far I have had no motherboard related issues. Any specific areas I should test? I know some have complained of poor audio quality and other such issues. I have not seen this. Any details on when you see the problems? And if they're related to the Ouch or Power alarm errors, I too get those. I'm starting another thread for those. One odd thing about the cpuinfo below. It reports the system as being a 533MHz system, when it's a 1.2GHz system (133 x 9). Even the BogoMIPS reflect the slower speed. I have no idea why yet... Tim Massey [EMAIL PROTECTED] proc]# cat cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 9 model name : VIA Nehemiah stepping: 8 cpu MHz : 533.440 cache size : 64 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr cx8 mtrr pge cmov pat mmx fxsr sse rng rng_en ace ace_en bogomips: 1052.21 [EMAIL PROTECTED] proc]# cat interrupts CPU0 0: 51670125 XT-PIC timer 1:256 XT-PIC i8042 2: 0 XT-PIC cascade 5: 22232 XT-PIC yenta, eth0 7: 51573795 XT-PIC yenta, wctdm 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 12: 58 XT-PIC i8042 14: 12064 XT-PIC ide0 15: 464588 XT-PIC ide1 NMI: 0 ERR: 0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Major problems with TDM400 and specific telephones: suggestions?
Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors. The phone in question is what I would consider to be a good-quality GE two-line cordless telephone. Digium's guess is that it is putting power on the telephone line and the card doesn't like that. They have given me zero solution other than to use a different telephone. If this were a $25 garbage telephone I could understand. Or, if *any* other device had problems with it, I could understand. But this was a reasonably expensive, seemingly reasonably high quality telephone. It is also a telephone that I have used quite successfully not only on standard POTS lines, but also on a variety of ISDN NT-1's with zero problems. I don't mean that I've used this model. I've used this *phone* on at least 4 different brands and models of NT-1's, several different POTS lines and even an SPA-2000. Not one bit of problem. Yet the TDM400 card just chokes itself with Power alarm and Ouch errors. Does anyone have any idea of what I can do to try to correct this? Is there some sort of filter or adapter that I can use to condition the line for the TDM400 FXS modules? I'm handy with a soldering iron: if you've got an idea for a circuit, I'm game. I'm going to try experimenting with some caps and coils. Anyone been down this road yet? As an aside, why is it that just about *any* other device with an analog interface you can buy today more robust than the TDM cards? I've used countless different ISDN NT-1's without problems, from $100 cheapo models to $1000 high-end devices and tons in between and none have had problems like this. Now there's a ton of SIP gateway devices. They don't seem to have these issues. Why do the TDM cards? And most importantly, can an end user do anything about this? Tim Massey Hello! I have been struggling to get a TDM400 card working for some time now. I have a TDM400 with 3 FXS modules and 1 FXO module. Right now, the FXO is installed but not connected to anything, and the FXS are connected to a number of telephones, including an inexpensive Lucent standalone analog phone, a 2-line ATT standalone analog phone, a 2-line GE analog speakerphone with AC adapter, an inexpensive VTech cordless telephone with AC adapter, and a 2-line GE cordless phone, also with AC adapter. I am testing the board right now just by calling back and forth internally from extension to extension. As long as I do not have the GE cordless phone plugged in, there seems to be no problems so far. At least, I do not believe I have yet seen the problem when the GE is not plugged in. Also, if I do not pick up the GE cordless phone, I do not get any errors. However, if I take the GE cordless phone off hook or put the phone back on hook, I will get Ouch, part reset, quickly restoring reality (#) errors, as well as Power alarm on module #, resetting! errors. This does not happen every time I pick up or hang up the phone, but after a relatively short number of times (say, under 20 at the very most) I will get one or both of the errors. It seems that the Ouch errors are indexed from zero, and the Power alarm errors are indexed from one. The port that generates these errors seems to vary. While the port that the GE cordless phone is plugged into seems to appear a decent amount, it is far from consistent. For example, right now the cordless phone is in port 1. However, I've gotten two Ouch errors, one for 0 and one for 1, and I've gotten three Power alarm errors, two for module 2 and one for module 1. After an error, channels on the the board often become very staticy. I believe that this only happens after a Power alarm, not after an Ouch error. In fact, I am pretty (but not completely) sure that after an Ouch error the board (or at least individual channels that are reset) clears up. Also, not just the channel that have had an error is affected: sometimes (but not always) all of the channels are affected. Sometimes the dialtone can be heard through the static, sometimes not. Even when the dialtone can be heard, it does not respond to DTMF tones. This varies from channel to channel: for example, right now, the dialtone can be heard through the static on one channel, but the other two have louder static, and I cannot hear a dialtone. I don't think the static is drowning out the dialtone: I think it's plain not there. Also, if I leave those channels off hook, I do not get a busy signal. Or at least, I don't hear a busy signal... Exiting Asterisk does not affect this. The static stays the same. Of course, with Asterisk exited, there is no
[Asterisk-Users] Succes report for TDM400 and IBM Netfinity 5600
Hello! I just wanted to tell everyone that I have successfully used a TDM400 with an IBM Netfinity 5600 server. I used PCI Slot 3 (the first hot-swap PCI slot). I had a ServeRAID 3L controller in slot 1as well, which managed the server's array. Other than that, there was nothing extra installed. 2x667MHz PIII, 1GB RAM, 4 x 18GB hot-swap SCSI drives. The only problems I had were Ouch and Power alarm errors that I believe were related more to the telephone in use, rather than the TDM card itself. More on that in another e-mail. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Major problems with TDM400 and specific telephones: suggestions?
[EMAIL PROTECTED] wrote on 03/22/2005 03:56:22 PM: On March 22, 2005 03:08 pm, [EMAIL PROTECTED] wrote: The phone in question is what I would consider to be a good-quality GE two-line cordless telephone. Digium's guess is that it is putting power on the telephone line and the card doesn't like that. They have given me zero solution other than to use a different telephone. I have a Panasonic 900MHz digital cordless phone that also causes the TDM card to have fits. I've sent it to Digium to try and figure out what's going on, as every single other phone and fax (probably two dozen brands between the two) I have ever hooked up has worked just fine. This is not a normal thing and it may just be that the actual POTS system is able to handle their particular brand of yuck. I certainly don't blame Digium for this, but they have been more than willing to help me correct it, especially since I am willing to get the phone to them to test with since they seem to be unable to recreate it in their lab. My 5.whateverGHz Panasonic digital cordless phone works great, and my 900MHz non-digital (cheapass) cordless phone works great. I too have a cordless that works fine: a very inexpensive Vtech cordless phone. Maybe that's the problem: we're buying phones that are of *too* high quality? :) As I said, I've hooked up countless devices to the TDM cards and this particuar phone is the ONLY one I've had trouble with. It is perhaps a corner case in the TDM design, but as I said Mark has personally been more than willing to help fix this. I would believe the I'm unlucky enough to have the one phone that doesn't work if there weren't so many other users with similar problems. It's disturbing that with N=5 I get a phone that doesn't work. Maybe if N=500 I'd still have just that one phone. But right now, I'm sitting at 20% failure! :) As an electronics designer myself, I know how unbelievably frustrating it is to have a customer with an issue and not be able to recreate it myself such that a fix can be found. As a computer programmer, I can certainly sympathize. And unlike, say, a Nortel system, we're expecting to plug phones from scores of companies into the system without problems. I don't envy their problem set. But again, I don't seem to have these issues with other, similar, devices. And I do have to concur with Digium's support. I just got off the phone with someone at Digium about my issue and they have given me a couple of things to do. We'll see if it helps, but at this point I'm happier. I'll let the list know what the result is. At least they're trying to fix it, rather than leaving me with We don't know why it doesn't work, and we don't care... That's all I can ask for. The killer is I *so* *want* this to work. The TDM boards would really fill a need if they work. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium support quality: Excellent
Hello! I wanted to make sure that, in addition to my complaints, I make it very clear: Digium's support is excellent. The jury is still out on the usefulness of the TDM products. However, Digium has worked very hard to make sure that this issue is resolved. I actually got an e-mail from someone at Digium actually asking what they could do to make me happy! She even gave me alternatives to hopefully correct my problem! And she was patient and friendly! I nearly fell off my chair. If you have any doubts about buying Digium products, don't let lack of support stop you. They stand behind their products with both technical support and customer service. You can't really ask for more than that. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Any interest in Line Powered Amplifiers?
Hello! I have a cabinet full of Wilcom Enhanced Line Powered Amplifiers with Manual Balance, model ELPA-421V. I *believe* these were used for a bank of analog modems back in the mid-90's. They were removed from a suite when the old company moved out. Here's a URL: http://www.wilcominc.com/elpa421v.htm Does anyone have any interest in these? If so, please reply off-list. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: How to own a telephone number?
Hello! We are open to the possibility of changing our business telephone number shortly. This will most likely be necessary due to a physical move, changing providers and a few other reasons. However, we woud like this to be the *last* time we need to do this. Ever. No matter what. Is that possible? On the Internet, you get this power with domain names. We own our domain name, so even if we move around the world, change connections, change Internet providers, grow, shrink, etc. we keep the same domain name. This is a wonderful thing. Is there such a way to do this with a telephone number? Is it possible to own a telephone number, such that even if we change telephone providers or move from POTS to ISDN to T1 to VoIP and back a dozen times we can keep the same number? We would like to have this power with both a normal telephone number and a toll-free number. According to our current provider (SBC in Michigan), the only way we can keep our current number is to convert it into a virtual circuit for almost $30/month (basically the same cost as a real circuit), and then forward all calls from that line to another number. If the number is not local, we'd have to pay for long-distance to that new number. I know that with VoIP numbers we can move and change Internet connections, but if we change VoIP providers, we lose the number. With a combination of the two, we could buy a virtual circuit from SBC and forward it to a local VoIP number that might change if we changed providers, but that seems like a fairly expensive way of doing it. Is there an alternative? And what if we didn't want to use VoIP, but wanted to forward to a number that was long-distance? Obviously, that gets expensive! Also, we're currently looking into toll-free service, but the alternatives seem to be much the same. At least nobody is telling us if there is a way to lock in a certain number even if we change providers. They've all told us that the number we receive is theirs, and if we change providers we lose the number. I'm sure 1-800-Flowers, et. al. are not being held hostage like that... I would love to know what ideas you might have for getting a telephone number with the ability to stay with us even as the underlying infrastructure changes. Is this even possible? Thank you, Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: How to own a telephone number?
asterisk-users@lists.digium.com wrote on 02/03/2005 02:20:57 PM: [EMAIL PROTECTED] wrote: Also, we're currently looking into toll-free service, but the alternatives seem to be much the same. At least nobody is telling us if there is a way to lock in a certain number even if we change providers. They've all told us that the number we receive is theirs, and if we change providers we lose the number. I'm sure 1-800-Flowers, et. al. are not being held hostage like that... What you are seeing with these bargain providers is they have a clause in their contract that says they own the number, not you. It is a lock, and it ought to be illegal, but sadly, it's probably not. If you choose one of these companies that doesn't allow you to port or resporg your number out, that's your decision. Just ask when you get the toll-free if they do allow resporg's out, and have them show you the wording in their contract that confirms it. Thank you for the information. That is what I was looking for, and I have now found providers that allow the numbers to be moved. I would love to know what ideas you might have for getting a telephone number with the ability to stay with us even as the underlying infrastructure changes. Is this even possible? A normal (not tollfree) number, if assigned to you by a RBOC, or most CLECs belongs to you, and you can port it to any other carrier who services your area(assuming they allow port-in's). I doubt you'll find a LEC that will want to do you any better than what you've already seen with the call-forwarding, unless you have a significant amount of traffic and want to set up a point-to-point, frame, or other method of trunking the traffic. Nope, just a small 4-person consulting shop. Not enough volume to be interesting. OK, then. If a $30/month for a virtual circuit forwarded is as good as it gets, then that pays for 600 minutes of toll-free number time at $0.05/minute. On top of the fact that we would like a toll-free number anyway, it looks like there is almost no reason to keep a permanent local number. We'll just have a permanent toll-free number instead. Is providing the ability to assign numbers to people instead of to locations really that hard? Is it really so much easier for Internet domains to do it? Or is this just an oligarchy at work? :) Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] - Advice on NetFinity 5000 series
[EMAIL PROTECTED] wrote on 10/07/2004 03:02:41 PM: I have an opportunity to pick up a couple of NetFinity 5500's 4 way Xeon 550's w/ 2 gig RAM for very little $$$ I have seen this: http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg00719.html In it, there is a passing remark to the Digium cards having problems with NetFinity's. Can anyone here comment on whether this is still an issue with * 1.0? It'd be a bummer if compatibility is a showstopper here 'cause these are sweet servers at a sweet price. That was my message. I have a lot of Netfinity servers. The 3500's are old, they do not support PCI 2.2, and I've had problems with more than just Digium cards with those servers (including IBM ServeRAID adapters: you could not reboot them, you had to power cycle them). I have both Netfinity 5500M20's (which you are describing) and 5600's that I have used with Digium hardware successfully. Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Module nonsense (zaptel, wcfxs and wxfxo)
Hello! I've been playing with two pieces of hardware: a X100P and a TDM400P with an FXO and two FXS modules. I had been using just the TDM card; however, the TDM FXO module seems to hear things and answer the telephone for no reason, and I wanted to compare the results with an X100P card. If you want further details, I can give them to you, but suffice it to say that trying to work with both cards and both modules has been incredibly frustrating. Modules that won't load, or that load but don't work when you run Asterisk, or Asterisk segfaulting even though the modules *seem* to load properly... Am I the only person who finds the combination of seeminly awkward separation between modules, duplicated information (between zaptel.conf and zapata.conf), strange interactions and limitations between different modules for what would seem to be very similar hardware? This is not a rhetorical question, nor is it a shot at the developers. I'm really asking two things: am I alone in this? Or are my expectations too high? I would be thrilled to give you whatever information you might desire if you're interested in seeing what I'm talking about. I'm using a dirt-simple configuration: basically four-line configuration files, just the minimum necessary to make the hardware work. Even these configurations cause a dizzying variety of issues... Again, I'm not trying to blast anyone. I just wonder what I can do to improve my success! :) Mostly, I would like to hear how others are fairing, particularly with a variety of analog devices. Am I alone? Thank you, Tim Massey
Re: [Asterisk-Users] PCI 2.2 ??
[EMAIL PROTECTED] wrote on 05/28/2004 03:51:02 PM: Dear users: I have bought TDM04B card and it works in PCI 2.2 ver. slot. How can I check if specific mother board support PCI 2.2 ver. I do not have any documentation for that motherboard. The easiest way is to look at the chipset. All Intel chipsets before the 8xx series (810, 820, etc.) do *not* support PCI 2.2. That means most systems with Intel chipsets and CPU's under 800MHz will not support them. Tim Massey
Re: [Asterisk-Users] Important: The Asterisk Mailing list (newsubject)
[EMAIL PROTECTED] wrote on 03/20/2004 02:58:21 AM: You give too much credit to people, indeed. I cannot say about this list, but most lists I use have high corporate populations, where the users *have* to use mailers like Outlook or (cringe) Notes. For mailing list admins to expect users of these mailers to try and find the functions referred to in the article is ludicrous (and yes, I know I just said in a prior note use the function of your mailer, but I was referring to the standard Reply function -- if you have ever tried to use a mailing list with Lotus Notes you will bless the list admin who maintains the status-quo and munges Reply-to). It's odd: I use Lotus Notes, and while I prefer the current Reply-to action, having to click the Reply To All button right next to the Reply To button is not exactly a hardship... I do miss the way my old mailer (PM Mail 2000) prompted to either Reply to one or all, but Notes doesn't exactly make it hard.. Way off-topic, I know, but I had to defend Notes! :) Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has Nufone gone belly-up
[EMAIL PROTECTED] wrote on 01/26/2004 01:12:09 AM: You're right, Jeremy. I made up the whole thing. I went out of my way to concoct a story about how I wanted to do business with you, but was unable to figure out how on your website, so I called and left a message and didn't get a return. Yeah, whatever. For the record, I had exactly the same problem with exactly the same response. I left one message a day for several days with no response. When I e-mailed about it I was told that that was impossible: I couldn't have called, the number was not on the webpage. I then showed him the link. He said the information wasn't there. When I went back, he was right: it wasn't there. Funny, though, the Google cache still showed the phone number... I can't speak about NuFone's service. I never got that far. I'm now using a different VoIP provider. However, when a person tells me what I have done was *impossible*, and when I show him how I did it he *still* doesn't follow up, I don't need to do business with them. Obviously, it's still a problem. You would think that when multiple people tell you that they have tried to get in touch with you and the messages are falling on the floor, you would do something about it. I guess they're too busy improving their service to take on additional customers... I've held off writing this to the list, but this was just too much. Mr. McNamara replied to Mr. Baker with the same brisque statement he made to me several months ago. I found his reaction unpleasant then, and I find it disturbing that others are having the same problem months later. Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage
[EMAIL PROTECTED] wrote on 07/31/2003 12:52:10 PM: www.nufone.net is entirely Asterisk/IAX. You know, I've called them several times and left my telephone number to call back. I've never heard from them. You know, many people on the list raved about them. But for a company with a completely useless website they really don't do a very good job of getting back to someone... It's not like I'm not already taking a chance calling such a company as it is. Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Summary of VoIP options for Asterisk and request for more?
Hello! Well, so much for mailing me off-list: not a single person did! In other words, you've already seen the results of my request: The options are: Nufone.net Cost: 2.9 cents/min for both outgoing long distance and incoming 800 calls. Service is pre-paid. Advantages: Extremely satisfied customers. IAX support. Disadvantages: One of the worst websites I've ever seen for a company. (Well, at least it doesn't use flash...) However, customers say that the customer side of the website is very good. www.global-gateway.net Cost: between 2.55 and 1.9 cents/min for calls (according to a customer) Advantages: I don't know. I could find out nothing about this company. Their website didn't work for me at all in either Mozilla or IE: All links merely pointed to #... Disadvantages: A completely broken website, no US telephone numbers. Seeing as only one person mentioned Global Gateway, and several raved about Nufone, I guess that's the direction I'm going to look. Nufone's service seems more geared to business use: (800) number for incoming, etc. I was more looking for something for home: a local telephone number, a number of minutes for a small monthly cost, etc. I'm going to have to review my home usage to see if it makes sense. However, my use of VoIP at home is in preparation for using it at the office, so maybe it's the best way to go... Are there any other options that people are using? IAX termination, while nice, is not required: SIP would work too. I'd prefer to stay away from H.323, but if you're using Asterisk to talk H.323 with a VoIP provider, I'd love to hear about it... Thank you very much for the responses. I really appreciate your help. Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] questions
[EMAIL PROTECTED] wrote on 07/18/2003 06:11:16 PM: On Fri, 2003-07-18 at 17:05, CTI wrote: Does anybody developed Predictive Dialer using Asterisk/Digium PBX? There has been talk about how to do this, but I don't remember anyone announcing it as either done, or open sourced. Can we brutally beat to the point of death anyone who actually *does* complete such a project? Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware Vendors
[EMAIL PROTECTED] wrote on 07/14/2003 12:37:33 PM: My fantasy machine for this purpose would be along the lines of a mini-itx system with external power supply, dual Ethernet interfaces on board, and one PCI slot available. If it had one real serial port on it, that would be great too. Am I dreaming, or does it exist for a reasonable price? I would be willing to go the 500 MHz 1 GHz range. Something without a fan would be really nice. Im basically looking for a system that someone out there is stamping out in quantities and isnt too outrageous in price. Does it exist, and if so who sells it? www.caseoutlet.com Via Eden 533MHz processor, no fans whatsoever. Runs like a PII 400MHz. They have cases that have 2 PCI slots. That's the biggest limitation: lack of PCI slots. We use these to sell Linux-based firewall computers for clients. They have run for well over a year with exactly zero crashes. With no moving parts (not even hard drives: we use DOM for the firewalls), there isn't a lot to go wrong. Having said all of that, I don't think they'll make good Asterisk boxes. 2 PCI slots isn't much and 400MHz PII-type performance isn't great (though you can get 750MHz or so of PIII performance from the new 1GHz CPU's if you don't mind a CPU fan). But if you can live with that, they're very nice. Don't forget to target the i586 architecture. The VIA CPU's don't have an instruction (CMOV? CMPXCHNG? something like that) that the Intels do and that CGG uses with an i686 target. Unfortunately, the VIA gets detected as an i686... Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
