Re: [Asterisk-Users] Red Alarm on X100P
3) It could be the motherboard. We're on the cheap here and used available components to make our server...worked fine with two x100's, then the boss wanted another line. Once I got the damn thing to accept them all on different IRQs, two would red alarm nearly every night at random times. Calling in on those lines, though the calls wouldn't answer, would clear the alarms. After trying various * and zaptel versions, I replaced the mobo with an older dual P3 I had. Either the slots were getting more power or something else in the mix fixed it. Don't care - it works and I have other things to worry about. Best, Bob Lyle Giese wrote: 1) it could be the x100p. Have you tried merely disconnecting the phone line and plugging it back in? 2) it could be the phone line connected to the x100p. A red alarm is an indicator of the co talk battery is missing on the line jack. Lyle - Original Message - From: "Mark C. Thomas" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, September 22, 2004 6:56 AM Subject: [Asterisk-Users] Red Alarm on X100P Hi, I have the following config, which I can elaborate on if necessary: TDM400P REV E/F X100P (X101P) PII-450 Linux version 2.6.8-gentoo-r3 gcc version 3.3.4 20040623 Asterisk CVS-HEAD-09/05/04-09:28:57 Last night I had called into this system and was talking to someone when the line on my side went dead, then I got dial tone. The remote extention just got silence the n then gave up after a minute and hungup. They then tried to call me back and got a fast busy. Now the strange part, I tried to call them, it would ring about 8 times, then I got DTMF tones like someone was dialing. They never heard any rings. I tried several times with the same result. I unloaded and reloaded asterisk and got the same result. I did a "show channels" and got no output. I ended up unloading and reloading wcfxo which finally cleared the problem. This type of thing happends about once every 2 weeks. I usually have them reboot as I am not available. Is there anything else we can do to try and narrow this down? I have been through several cvs updates and the problem persists. Could it be the x100p card? ___ 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] Buffers and Caches and realizations (Was: 1.0_stable ....)
Re: Memory: The cool thing is Linux can just discard the cached entries when a application needs real RAM. Don't worry about your RAM usage until you see swap climbing and/or the buffers and cache dropping down to near zero. Yes, yes, I knew about caches versus HD access, but I didn't realize what types of cache there were and what their uses were. It struck me when you mentioned "buffers and caches," as they are labeled in top, that a buffer shouldn't stay full, and why should my bloody system have 150M of unflushed buffers? But, of course the labels meant "buffer cache." I went looking for some more info on this and found this, if anyone's interested: http://www.linux-tutorial.info/cgi-bin/display.pl?310&0&317&0&3 I _am_ curious how much overhead is involved in deciding what to dump, dumping the page, then allocation, and if that could have any effect on time critical applications with a recent linux kernel. I'm betting not much if at all, but I'll read up on it when I have the time. [OT]: Maybe if this subject hadn't been covered 3 times in less weeks. You seem to first have missed the previous comments on this subject, and second you could have overlooked a one line non personal vent when there was a couple of paragraphs that explained clearly what you needed to know. [You know Critch, I have to apologize - my impression of you, beyond the time you bummed our floorspace at DragonCon, was of a self-important suffer-not-the-newb kind of guy. It was a lot of that kind of sniping that led me to blow off the NLUG list a while back, and my daily runs through the * list traffic---more skimming really...not enough time for this and work too--- seemed to confirm it. I just went looking through the archive, though, and noticed far fewer harping messages from you than I thought. I think I fixated on the "read more before you ask again" posts because you tend to more and top post less making them more quickly readable, and because I pay more attention to the general questions. My "sanctimonious bullshit" remark was meant to encompass _all_ the harp that I thought you enganged in, which turns out to be much less than I thought. Fancy that. By the way, do you know where that mythical kiss-my-ass list is hosted, and how much they charge?] Bob ___ 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] 1.0_stable is or isn't? (Was: No ringing tone...)
Steven Critchfield wrote: On Sat, 2004-04-10 at 11:51, Bob Klepfer wrote: (I *have* noticed RAM almost completely filled, but no swap used...a reboot freed a bunch and I think that fixed some issues. We're a small company and restarting * or rebooting the server isn't that big a deal.) Once again we must teach a newbie about memory usage and the tools they use to check it. If you want to clear up a perceived point of misconception, please do Critch, but will you spare us the sanctimonious bullshit? ___ 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] 1.0_stable is or isn't? (Was: No ringing tone...)
Brian Cuthie wrote: What version of the Asterisk code are you running? 1_0 stable is definitely broken wrt ringback, and the latest stuff seems really broken in all kinds of ways. After seeing that others were having similar problems, and that someone had solved many of them by rolling back to the CVS version from 3/5, I tried the same and things are working marvelously (well, mostly). I've been swamped at work and heven't been able to keep up with the version discussions or monitor asterisk-cvs closely. Could you qualify your statement above about 1-0_stable being broken? I'm running 1.0 stable (CVS-03/20/04-22:33:52) here at work and have noticed faxing over SIP much more stable, but a couple of momentary dropouts on outside calls (GS bt101 -> x100p POTS), usually after silence in the conversation. (I *have* noticed RAM almost completely filled, but no swap used...a reboot freed a bunch and I think that fixed some issues. We're a small company and restarting * or rebooting the server isn't that big a deal.) Bob ___ 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] Spring VON Wrap Up
Mark Messmore, Technical Support, University Telcom Inc. wrote: K...maybe this was stated earlier in the conversation...but what's the deal with the phone? Or was this phone just being carried around by everyone and ripped apart? Old Bell Phone + IAXy + 802.11b card + Batteries = Homemade WiSIP, IIRC ___ 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] Asterisk on FreeBSD
Olle E. Johansson wrote: Richard Airlie wrote: On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote: Richard Airlie wrote: At this stage, and someone please confirm that Asterisk is really working on FreeBSD ? :) Yes, it's working with some limitations. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd Thanks for that, good to know. And now leads me to ask... why should my SIP softphones be unable to register? They are on the same subnet as asterisk. If i have sip debug turned on, shouldn't I at least be seeing some action on the Asterisk console when they try to register? Turn on SIP debug and you'll be able to see what happens. Check with "sockstat -l" if Asterisk is listening to port 5060. Also, make sure you start asterisk with a lot of -v to get debug output. Since you see no SIP traffic with SIP debug on, is ipfw blocking SIP? Bob ___ 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] LARGE BREASTS Handoff back to * from * via IAX?
Andrew Kohlsmith wrote: mmm... I just wondered, since it's very likely that most people ended up deleting it *because* of the subject line. .. so it probably wont help ... well it might... I don't know -- It seems that plain English words are not in spam at all these days... It would have read "L AGR3 B*REAs3T5" or something.. You mean like "Best Web Hosting Service" or "Get Office Space Quotes" ? :-) ___ 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] Is asterisks the best for a simple DTMF response system?
Bryce Nesbitt (mailing list account) wrote: I received a recommendation to check out Asterisk, as a platform to host a simple DTMF response system, something like: Setup up VoIP endpoint on Linux/FreeBSD system Answer incoming VoIP phone calls User enters 100#, perl script plays back "foo" User enters 101#, perl script plays back "fum" User enters 102#, perl script looks up something in database, converts to text with festival, speaks it. 100, 101 are built in, no perl needed, 102 may require a short script. How would one get started, using Asterisks on this project, Read. http://www.voip-info.org/wiki-Asterisk Also the config files that come with it. It takes some study time to absorb. Or hire a consultant. and is Asterisks the best option? It's an excellent option, though pretty underutilized for your application. Is it really good enough for a high volume (though sub carrier-grade) solution? Yes ___ 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] X100P fails to detect user hung up
Try calling application "Hangup" at the ends of the extension chains. Works for me. Bob [EMAIL PROTECTED] wrote: Ron, It is a multi-reported problem, yet no resolution. I would suggest it is a bug. I have had intermittent success with POTS provided by AllTel in Texas. My opinion, you're SOL and there is very little you can do. I keep hoping that someone at digium will pick up on this and look at the hardware design etc. BTW, I tried kewlstart, loopstart etc. and it doesn't make any difference. As I said, it's intermittent on POTS, and it's constant on my ISDN fxs channels. Cheers, Willy - Original Message Follows - I am using the wildcard X100P with *. PSTN line comes in to the FXO port of this card. Everything works fine most of the time. However, occasionally Asterisk doesn't seem to be able to detect the user has hung up and therefore tie up the line for quite a long time. Does anyone know if there's anything I can do to fix this problem? thanks Ron ___ 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] MOH: Copyright issues?
Alex Volkov wrote: AFAIK, in US the copyright expires 25 years after the original copyright holder (author, recording artist, but not sure about an assignee) dies, or after ~70 years from the date of creation (in cases where a corporation holds a copyright for sure), but do not hold your breath, as the companies like Disney constantly lobby to extend this period, otherwise you would certainly see Mickey Mouse cartoons in public domain by now. As far as royalties are concerned, I suppose MOH in US for some company could be considered on par with a bar, which translates to pennies per played song, as long as no more than ~100 people are listening to it at once. But please do not take this a as sound law advice, as I am no lawyer ;-). Cheers! Alex. Unfortunately, it's much more heinous: 70 years *from the death of the last remaining creator*, if not a work for hire, anonymous, or pseudonymonous work. If it is a work for hire, 95 from first publication, or 120 from creation, whichever ends first. Whether is the work was in the first or second period of copyright (first 28 years) before 1978 changes some things.I don't know - it takes a lawyer or a bought-and-paid-for politian to read this crap. Too many words, not enough equations :) http://www.copyright.gov/title17/92chap3.html#302 Bob ___ 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] Newbie Start Question
Mamadou Lamine KA wrote:
I am very new to Asterisk. I want to set up a PBX and an IVR server with it.
I have a wildcard X100P and a TDM400P on my RedHat box.
I have installed Asterisk and the devices and everything seems OK. ( Asterisk Ready )
Now I want to launch the Demo context in /etc/asterisk/extensions.conf so that when a call comes it is directed on that context. How shall I proceed?
Make sure the X100p channel is numbered consistantly between
/etc/zapata.conf and /etc/asterisk/zaptel.conf, is being signalled with
fxs signalling, and has a context ("default", perhaps) defined in
extensions.conf where include => demo is included, assuming you didn't
change your extensions.conf much from the sample. Channels are numbered
based on what module loaded first. In your case, loading wcfxo then
wcfxs would lead to channel 1 = x100p card, and channels 2-5 = tdm400p.
I have of course read the Asterisk handbook but it is too the theorical to me. Could someone tell me where i can find exact informations on how to set up and how to use IVR server with Asterisk.
Keep reading and absorb the .conf files. It will gel at some point.
Also see http://digium.com/downloads/configuring_zaptel.pdf
Best regards,
Bob
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Re: [Asterisk-Users] Fax Detection on X100Ps (Fixed!...I think)
Rich Adamson wrote: Bob, Help the rest of us out now and summarize the various *.conf entries that you have working. Might even start a new posting with a subject that will help everyone find your samples. Rich I was planning to, Rich, as soon as I've finished the long delayed rollout here, if you can call 11 phones and a fax a rollout. I honestly can't spot the change I made to get the card to recognize the fax tones, but I'll post and/or wiki-ize my configs soon. Bob ___ 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] Fax Detection on X100Ps (Fixed!...I think)
Replying to my own email here... Bob Klepfer wrote: Jonathan Biggs wrote: exten => fax,1,Dial(SIP/ata4fax) ; [1] Faxing via SIP? Does that even work? Faxing works for me but it is via ZAP. When I started I saw no obvious signs that it doesn't. I've seen several references to a SIP channel in example fax exten lines, but documentation is scarce. See below. I'm watching the asterisk-r off an asterisk started with -vvvc, and I see nothing like that message. If anyone wants to see other configs, let me know. I just realized I haven't tested both x100p cardsthat's next...after work when I get a chance to test. Bob Um, ok. So it works now. Redirect message is there, and the fax answers. I didn't even try the second card, but I did try the "d" option to Dial (still looking for this documentation.).and now 2 minutes later I just tried without "d" and, as I expected, it still works without it. What actually changed to make this work, I couldn't tell ya. Maybe it was that static spark between me and the shelving where the server sits. :) Thanks anyway, Bob ___ 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] Fax Detection on X100Ps
Jonathan Biggs wrote: exten => fax,1,Dial(SIP/ata4fax) ; [1] Faxing via SIP? Does that even work? Faxing works for me but it is via ZAP. When I started I saw no obvious signs that it doesn't. I've seen several references to a SIP channel in example fax exten lines, but documentation is scarce. I do get the messages Fax detected redirecting to Fax extension. Which you should get irregardless of SIP. And this is my quandry.. How are you testing this. Asterisk listens for the Fax Tone, I see you are answering the line which is a must, Background should detect this I think.. yup. First step is to get the Fax detected by incoming That's the plan. Redirecting Zap/1-1 to fax extension This message does not show up in my log but DOES show up in my asterisk -r session or a redirection of asterisk -vvvc stdout to a file I'm watching the asterisk-r off an asterisk started with -vvvc, and I see nothing like that message. If anyone wants to see other configs, let me know. I just realized I haven't tested both x100p cardsthat's next...after work when I get a chance to test. Bob ___ 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] Fax Detection on X100Ps
I hate to add to the broken record-like melange of "my fax won't work" messages, but everything I've tried with all I could learn from the archives has not yet worked to get my fax machine (an HP combo tupperware tub) to receive a fax. In the combo's defense, I can't verify that the incoming fax tones are even detected on the * server. (is -c enough v's to see "Fax detected" messages from the zap channel?). The fax extension is never dialed and the incoming fax gets dumped to my timeout extension. Can anyone spot the bonehead move I made somewhere? The setup: * Server with 2 X100p cards handling our two incoming analog lines. (Gentoo, if it matters---2.4.22-r7 sources) * SIP only (so far) internally---no FXS cards * An HP combo fax/copier/printer/dish cleaner connected to a GS Handytone 286 ATA * Incoming and outgoing voice calls are routed fine * Although I didn't have another machine to fax TO, the HP dials out and spews CNG tones * zap context is "default", "default" context contains fax extension * on reload, status includes: "-- Added extension 'fax' priority 1 to default" Version: (cvs'd last night and built, previous version cvs'd and built mid-Feb) === Asterisk CVS-01/27/04-23:36:00 built by [EMAIL PROTECTED] on a i686 running Linux Config files: == in extensions.conf: -- [default] ; exten => s,1,Answer exten => s,2,Ringing exten => s,3,Wait,2 exten => s,4,Background(photonx/welcome) exten => s,5,Background(photonx/choose) exten => s,6,Background(photonx/directory) exten => fax,1,Dial(SIP/ata4fax) ; [1] exten => fax,2,Hangup() ; --- [1] (Awaiting an opportunity to test "d" option mentioned on list (documentation for this?)) === in sip.conf: --- [general] port = 5060 bindaddr = 0.0.0.0 context = internal [ata4fax] type=friend username=ata4fax secret=atafax context=internal host=dynamic disallow=all allow=ulaw dmtfmode=inband ; (Was "info": neither an issue, I think) canreinvite=no --- === in zapata.conf: --- [channels] ; ; generally stock - no calling options on our lines language=en context=default usecallerid=no hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=yes ;relaxdtmf=yes rxgain=10.0 txgain=5.0 group=1 callgroup=1 pickupgroup=1 immediate=yes callerid=asreceived amaflags=documentation ;busydetect=yes ;busycount=4 signalling=fxs_ks channel => 1-2 --- Best regards, Bob ___ 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] Switch brands, speeds, etc.
The short of it: In light of the recent Netgear posts, I'm just curious if anyone has preferences for brands of switches - we're wiring a parallel network of 10BaseT over existing cat3 for the IP phones in our office space. The long of it: --- Our setup: * Office of <10 people spread out in 2000-3000 sq.ft. * Space previously used as computer learning center, chock full of cat-3 and multiple rj-45 jacks per wall plate. * We're rewiring anyway - company growth + lack of planning has led to switches and hubs strung everywhere * I've convinced the boss to let me implement an asterisk server, replacing the unholy phone concoction we have now * No external VOIPat least not yet. * MUCH data flying back and forth from computers in labs to offices and vice versa So we were thinking of using some of the existing cat3 for just the IP phones and stringing some cat5e alongside for intranet. Buy a cheap 10BaseT switch (SmallDog has a refurbished Asante 5324 24-port cheap) for the cat3 lines and feed that to our * server's eth1. We're geeks, but not really networking geeks, so I thought I'd ask the list populace at large if they had comments/recommendations. Best, Bob Klepfer Photon-X, Inc. ___ 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] Whats wrong with dialplan?
Chris Lee wrote: I am having problems with my dial plan, please help me locate the problem: In the following dialplan, I am not able to press 8 to get to voicemail main while the 3000 mailbox unavailable message is being read in the background. What am I doing wrong? [well-road] ;includes include => voicmail access include => extensions include => no match voicemail is misspelled - would that do it? Bob ___ 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] Incoming Voice/Fax Discrimination?
I'm evaluating * to replace the crap set of peered "smart" phones we have now in our small office, but I haven't been able to find out about this anywhere yet: I need to know if * can discriminate _incoming_ FAX calls on a voice line and route them to a specific extension? We have a little standalone box to do this now, but only for one line, and if that line is busy---we have two with rollover--the fax doesn't get picked up. I'd obviously like to tell the boss that * can handle this function, too. Best, Bob ___ 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
