RE: [Asterisk-Users] TDM card and voicemail volume
> -Original Message- > From: Steve Prior [mailto:[EMAIL PROTECTED] > Here is the text of the last 2 bug comments by MikeJ (who I > would assume > closed the bug). I think there are three issues here: 1. The bug was originally filed as a "feature request" for a feature that would have been a work-around, at best. The actual source of the problem wasn't narrowed down until later. It probably should have been filed as a bug report, instead. Unfortunately, I fear that trying to file one now will probably just result in it being marked as a duplicate of the closed feature request. 2. I believe there are quite possibly two seperate bugs conflated in that one item. There's the recording format problem (compressed formats are at -6 or -10 dB compared to uncompressed) and possibly also a TDM-specific recording volume problem. 3. The people who are affected are not the people who are capable of fixing it, and the people who are capable of fixing it are apparently not affected enough by it to care. ___ 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
> -Original Message- > From: Adam Robins [mailto:[EMAIL PROTECTED] > 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. This is an acceptable work-around if you're just doing voicemail and IVR. It may cause echo or excessive volume levels if you're also doing regular calls, though. ___ 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] Comedian Mail User Setup Prompts
> -Original Message- > From: Adam Robins [mailto:[EMAIL PROTECTED] > I have a user who goes into Comedian Mail for the first time and goes > thru the initial setup, changes password, records name, etc. > Problem is > that every time he calls in, it thinks that it's his first time and > keeps reprompting him. His password change is reflected in > voicemail.conf. Others do not have this problem. Last time I had this issue, it was because someone was trying to set their PIN to their extension number! Sounds like you already checked for that, though. ___ 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] LiveVoip is Bankrupt
> -Original Message- > From: Matt Riddell [mailto:[EMAIL PROTECTED] > Heh, go easy on the guy, he probably hasn't got threads and > has to read > every topic just to get to the topics he likes. > > :D > > X-Mailer: Microsoft Outlook, Build 10.0.6626 Right click on the message list heading, Customize Current View..., Group By..., Conversation. ___ 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
> -Original Message- > From: Adam Robins [mailto:[EMAIL PROTECTED] > I saw some conversation about this in the archives, but nothing > definitive. > > If a call comes in over a CO line via the TDM400P, the Comedian Mail > recording volume is so low it's inaudible. Calls coming in via SIP or > IAX do not have this problem. > > Does anyone have any information on this issue? Workaround: Use the uncompressed WAV format. This is good for about +10 dB compared to the compressed formats, for some bizarre reason. It's not just voicemail, recordings made with Record() are affected, too. There was a bug filed about this. It was marked as a wontfix, as I recall, because no one would step forward with code or $$$. ___ 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] Re: Re: Digium Website Update: Asterisk Busi ness Edition
> -Original Message- > From: Esben Stien [mailto:[EMAIL PROTECTED] > The other problem is the issue that free software developers are > mostly (in my experience) not happy with the fact that their code > would be used in proprietary software. It conflicts with the whole > religion of free software. Well, yeah, that's the whole problem, isn't it? You can't follow the "religion of free software" and still run a company that pays the bills. You have to compromise somewhere. Either you go out of business, or you tick off some of the open source purists. Interesting perspective on this from Forbes: http://www.forbes.com/technology/2005/05/26/cz_dl_0526linux.html ___ 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] Re: Asterisk forking, Was: Digium Website Up date:Asterisk Business Edition
> -Original Message- > From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] > On Monday 13 June 2005 12:38, The VoIP Connection wrote: > > This is a very interesting converation, but it seems like > the BIZ forum > > might be more appropriate... > > How on earth is this a business-related discussion? -dev > would have been my > guess. :-) Maybe we need an anti-biz list for this kind of thing. ;) ___ 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] Voicemail and MS Exchange Synchronization
> -Original Message- > From: C F [mailto:[EMAIL PROTECTED] > Sent: Saturday, June 11, 2005 11:28 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Voicemail and MS Exchange > Synchronization > > > On 6/10/05, Dean Collins <[EMAIL PROTECTED]> wrote: > > Actually I think that has changed to 75gb now (or about to change). > > > Really? any links to support that? Since when is Micro$oft so easy on > giving up on licensing fees? I'm curious, too. If this is true it might save us a lot of pain, upgrade wise. We've been looking at moving away from Exchange entirely because of that damn 16-gig limit, and Exchange Enterprise Edition is just too expensive. ___ 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] Re: Voicemail and MS Exchange Synchronizatio n
> -Original Message- > From: Iassen Hristov [mailto:[EMAIL PROTECTED] > Does this matter? All we are saying is that Exchange supports > IMAP and we > would use IMAP as the protocol to delete the message from the user's > mailbox. How does the user access his mailbox is his choice. I think two threads of discussion got crossed. Somewhere along the line someone brought up the idea of having Asterisk act like an IMAP *server* where people could retrieve their voicemails. ___ 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] Voicemail and MS Exchange Synchronization
> -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > That's a concern, especially since bugs in current Asterisk versions > require > you to use uncompressed WAV files to get acceptable volume levels. > However, > this *is* a common configuration for other products. > > ) I do not worry about this. It is 'only' storage space. Yes, but Exchange storage space is expensive. The "small business" and "standard" versions of Exchange have a 16 gigabyte storage limit. If you want to exceed that, you have to shell out about $4,000 extra for the "enterprise edition." ___ 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] Re: Voicemail and MS Exchange Synchronizatio n
> -Original Message- > From: Iassen Hristov [mailto:[EMAIL PROTECTED] Dumb, hacky idea...but just so crazy it might work: Have Asterisk include a read receipt request when sending the voice mail message. Write a script, triggered from a sendmail alias or .forward file, that will parse the incoming receipts and handle the message deletion. Bonus points: When someone listens to the message on the voicemail server, send an Outlook message retraction request. ___ 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] Voicemail and MS Exchange
> -Original Message- > From: magnus [mailto:[EMAIL PROTECTED] > >From my perspective, not sure I would want Exchange (Which > is difficult enough to manage) to be cluttered up with > potentially large voicemail files, That's a concern, especially since bugs in current Asterisk versions require you to use uncompressed WAV files to get acceptable volume levels. However, this *is* a common configuration for other products. We used to have a CallXpress system that used Exchange as a message store. It stored voice messages in people's Exchange mailboxes, and could even read email messages over the phone via text-to-speech. The interface with Exchange was kind of kludgy, though, and not entirely reliable. It actually used a copy of Outlook on the voicemail server to talk to Exchange. > I would have thought that most Exchange clients are most likely to be > Outlook based, who could use pst & Imap (Or pop3 if asterisk > could auto > forward and then delete voice mail) to retrieve voicemail via > email without > having to worry about central Exchange issues. IMAP is no good. Outlook, at least in older versions, cannot handle both an IMAP account and an Exchange account at the same time. (They can do POP3 and Exchange together, though.) A voicemail app that used an IMAP server as its message store would still be a nice feature, though. It might even work with Exchange, which can act as an IMAP server. ___ 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
> -Original Message- > From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED] > I did that once on a cheap linejack card. Took a week to get > the smell > out of the office, and the bright orange from inside the server was > quite interesting :) Only took 1 second to start a small flame going, > but fortunately I cought it quick. Reminds me of when I smoked cheap a sound card connecting it to the tape output of a tube amp. The sound card apparently had no DC blocking capacitor on its input, and the tube amp had some DC on its output... > > I wonder if the zaptel cards have any kind of protection from > this sort of thing... No, they don't. Someone mentioned damaging one this way just a couple weeks ago. ___ 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] Two TDM04 with Poweredge
> -Original Message- > From: Tom Hayden [mailto:[EMAIL PROTECTED] > Has anyone on this list succesfully managed to get two (or more) TDM04 > (with four FXO each) working on a Dell PowerEdge server? If so, which > model? Was it a hassle? I've got a PowerEdge 800 tower server with two of them. Only five FXO modules right now, though. It mostly works. When I insert the driver I get an NMI, but that appears to be harmless. I have to unload and reload the drivers once a week or so, otherwise the FXO modules tend to eventually stop responding. I haven't had any audio quality or interrupt problems, though. The system gets the job done, but I can't wholeheartedly recommend these cards. If I had to do it all over again, I'd consider some other method. I'm not sure if anything else would be practical, though. A T1 card plus channel bank is kind of cost prohibitive for such a small installation. I've heard good things about the Sipura gateways, but I'm interfacing to a PBX and need the ability to flash the line for transfers, and I think Flash() is Zap-specific. ___ 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: carrying a router, firewall, switch, ser ver, some phones with me on flight to Europe
> -Original Message- > From: Giles Coochey [mailto:[EMAIL PROTECTED] > * While most PC PSUs these days are 100-240V, and they seem to have no > problems operating both in Europe and the US. UPSs are different, > however, they are almost universally either 110V or 240V only, and > there's not even a switch to switch between the two voltages. APC will > sell you either a US or a EU version, and usually only if they're > shipping to the destination. > > * Just a small UPS will probably do your baggage allowance in as well Yes. The company I work for occasionally ships configured PCs (being used as industrial controllers) to European countries. The PCs themselves are no problem, but if a UPS is required we always have them buy it locally. They're hard to get for European power standards in the U.S. (I've tried), and they're heavy and expensive to ship. Don't put anything in your checked bags you can't afford to lose. One of my friends is a travelling technician and regularly checks a bag of tools. It almost always arrives with something missing. He's lost three Leatherman tools. The TSA won't allow you to lock bags anymore, and the TSA inspectors and/or baggage handlers apparently have sticky fingers. If it's expensive to replace and too big to put in a carry on, consider shipping it to your destination instead of putting it in your luggage. ___ 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] What do you name yours
> -Original Message- > From: Ken Diliberto [mailto:[EMAIL PROTECTED] > On the surface, naming may sound trivial. When you're > dealing with users > and people paying your salary/consulting fees, it's not. > Offending the > wrong person because you named the server "Nag", "Chatterbox", > "ETPhoneHome", etc. can be very costly. > > You could use some of the following: > > PhoneSystem > PhoneManager > PhoneVoiceMail > CompanyPhone I named mine "dialtone". I try to give machines here names that relate to their purpose, but ones that aren't *too* software-specific, because renaming machines can be a pain. (For example, I'm reluctant to give names like "asterisk" or "squid" to machines, because what if I switch to some other package later?) I also came in when some machine names had already been established, so I didn't really have the option of using a cute, consistent scheme like some places do. So, for example, we have a proxy server named "gatekeeper", firewalls named "drawbridge" and "portcullis", and an AutoDesk Inventor Vault server named "edison". My users seem to be able to remember the names when they need to, which is the important thing. ___ 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] Interrupting voicemail with "*", dropping to "a" extension. Does it work?
> -Original Message- > From: John Lange [mailto:[EMAIL PROTECTED] > So to make "*" work you must have: > > [voicemail] > exten => a,1,VoiceMailMain() > > Does this make sense to anyone? I think it uses whatever context the current voice mailbox is defined in, in voicemail.conf. ___ 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] Newer Dell Servers + TDM card
I find this game kind of infuriating. If you have problems, they tell you to buy a different motherboard. But they don't supply a list of "approved" ones that they'll support. > -Original Message- > From: Matt Schulte [mailto:[EMAIL PROTECTED] > Sent: Thursday, May 05, 2005 12:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card > > > I just called them and they told me to quit buying dell. I > explained the problem in detail (déjà vu), and still no dice. > Is there a bug number I can reference for this? Did this fix > the problem? > > Matt > > > -Original Message- > From: Mark Phillips [mailto:[EMAIL PROTECTED] > Sent: Wednesday, May 04, 2005 12:42 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Newer Dell Servers + TDM card > > > Folks, > > This is a firmware bug in the TDMxxx and TExxx cards that Digium has > recently fixed. > > I did an "advanced replacement" for mine which involved me buying > another one and them refunding me when they got my old one back. > > Get onto their tech support. > > Mark > > Matt Schulte wrote: > > Is this with the TDM400P card right? > > > > -Original Message- > > From: David Brodbeck [mailto:[EMAIL PROTECTED] > > Sent: Monday, May 02, 2005 2:35 PM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card > > > > > > > >>-Original Message- > >>From: Matt Schulte [mailto:[EMAIL PROTECTED] > > > > > >>Really, how long does it take to "recover"? Mine just totally locks. > > > > > > No time at all. The only reason I know an NMI occurs is the front > > panel light, and the "Dazed and confused, but trying to continue" > > message from the kernel. I'm using a Dell PowerEdge 800. > > ___ > > 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 > > > ___ > 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 > ___ 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] voicemail volume with sipura 3000
> -Original Message- > From: Tim Connolly [mailto:[EMAIL PROTECTED] > Dunno.. Guess somewhere in the translation it gets amplified to an > acceptable level. It seems to work though. The issue appears to be that, for some reason, files recorded as GSM or WAV49 are encoded at a lower volume level than uncompressed WAV. This is true whether they were recorded in VoiceMail() or Record(). No one seems to know why, and the bug reporting it has been stuck at the "Feature Request" priority for months. ___ 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] Newer Dell Servers + TDM card
> -Original Message- > From: Mark Phillips [mailto:[EMAIL PROTECTED] > Sent: Wednesday, May 04, 2005 1:42 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Newer Dell Servers + TDM card > > > Folks, > > This is a firmware bug in the TDMxxx and TExxx cards that Digium has > recently fixed. > > I did an "advanced replacement" for mine which involved me buying > another one and them refunding me when they got my old one back. > > Get onto their tech support. Thanks, Mark, I'll give it a try. I hadn't bothered with their tech support up to this point, because the impression I'd gotten on this list was that they'd just tell me to go buy a different motherboard. ___ 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] Newer Dell Servers + TDM card
Yes. Actually, I have two of them in that machine. > -Original Message- > From: Matt Schulte [mailto:[EMAIL PROTECTED] > Sent: Wednesday, May 04, 2005 10:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card > > > Is this with the TDM400P card right? > > -Original Message- > From: David Brodbeck [mailto:[EMAIL PROTECTED] > Sent: Monday, May 02, 2005 2:35 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card > > > > -Original Message- > > From: Matt Schulte [mailto:[EMAIL PROTECTED] > > > Really, how long does it take to "recover"? Mine just totally locks. > > No time at all. The only reason I know an NMI occurs is the > front panel > light, and the "Dazed and confused, but trying to continue" > message from > the kernel. I'm using a Dell PowerEdge 800. > ___ > 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 > ___ 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] Newer Dell Servers + TDM card
> -Original Message- > From: Matt Schulte [mailto:[EMAIL PROTECTED] > Really, how long does it take to "recover"? Mine just totally locks. No time at all. The only reason I know an NMI occurs is the front panel light, and the "Dazed and confused, but trying to continue" message from the kernel. I'm using a Dell PowerEdge 800. ___ 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] Newer Dell Servers + TDM card
> -Original Message- > From: Matt Schulte [mailto:[EMAIL PROTECTED] > Has anyone ever been able to fix this NMI "power" issue that > the Dell's > have with the TDM cards? Basically locks the machine up when trying to > bring up the module. I get an NMI the first time I load the module, but the machine always recovers. Subsequent load/unload cycles don't trigger further NMIs. I'd like to know of any way to fix it, too, 'cause that orange flashing light is kind of annoying. ;) ___ 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] FXO lines on TDM04B not responding
-Original Message-From: Goutam Shaw [mailto:[EMAIL PROTECTED]Sent: Tuesday, April 19, 2005 11:22 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] FXO lines on TDM04B not responding I ran into the situation where 3 of the 4 lines on the FXO card stopped responding to the incoming call. I have 2 cards with a total of 8 FXO lines. A month ago we have replaced the old cards with the latest Digium X100M RevB. Before the card replacement the whole system used to get locked up but this time only 3 of the lines were not resonding on one card and the rest were fine. Digium guys donât say anything except reloading the driver and asterisk. However, in telephony as we all know this is not an acceptable solution. Is Digium HW is really bad.[David Brodbeck] I have mine automatically reload early on Sunday morning, when call volume is pretty much nonexistent, to get around this problem. I agree it sucks to have to do this. ___ 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] Toshiba CTX100 integration with PABX for two site
> -Original Message- > From: Henry Devito [mailto:[EMAIL PROTECTED] > I am currently working on the coding to provide D tone > disconnect. Keep us posted; I need this too. My * application is currently only IVR and voicemail, so I can work around it by setting the voice mail app to end a message after detecting silence, but that's kind of a hack. ___ 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 Revision question.
> -Original Message- > From: Rich Adamson [mailto:[EMAIL PROTECTED] > > As far as the issue with DC voltage on the POTS line only > > being 43.8 DC, my guess was that is just an issue with > > voltage drop on the line because of distance between me > > and the CO. > > No possible way. If everything is truly on hook, there isn't > any current draw and therefore no way for a voltage drop to > occur. Basic ohm's law. If he's not using a high-impedance voltmeter, the meter might be loading the circuit down. ___ 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 sought on how to automatically and sa fely reboot * box
> -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > I am trying to figure out a way to automatically shutdown and > reboot an * box safely on a set schedule. I have thought > about using a CRON job but I am a newbie when it comes to > setting up CRON jobs. I have googled for examples but I have > came up empty. Has anyone on the list setup anything similiar > to what I am wanting to do? You might look here: http://www.voip-info.org/wiki-Asterisk+automatic+daily+restart Neither of the examples does exactly what you want, but you could probably adapt them. ___ 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] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars
> -Original Message- > From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED] > I don't think the GPL obliges you to "give credit" to > anybody. In fact, I think that's a key difference between the GPL and the BSD license. ___ 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] x100p disconnect on "D" tone
> -Original Message- > From: Henry Devito [mailto:[EMAIL PROTECTED] > The zaptel driver has the 'D' Tone defined in the 'tones.h' > file I am trying to figure out what I can do with asterisk so it will > recognize that and do a HangUp. I need this too. Keep us posted. Right now I include it as an extension in all my IVR menus, and I hacked the "Directory" app to recognize it. I was going to hack VoiceMail, but couldn't figure out how. It'd be nice if there was a general way to handle this. ___ 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 becomes after one month unstabled
> -Original Message- > From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] > 1. The Zapta card (2 FXS, 2 FXO) suddenly does not like one phone. It > simple does not supply with a dial tone. You cannot dial. You > can reach > it, better say, you can dial it, it rings, but no sound. > > I reloaded and even restarted * without success. Than I removed the > module and inserted it again, ... restarted * and it works again. Yup, mine does that too after a couple of weeks. I have a script to automatically unload and reload Asterisk and the modules. The script I use is in the Wiki, near the bottom of this page: http://www.voip-info.org/wiki-Asterisk+automatic+daily+restart That's a band-aid, of course, but until Digim tells us all how to fix their cards, that's about the best we can do. ___ 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] Zap (analog line) and volume
> -Original Message- > From: Rich Adamson [mailto:[EMAIL PROTECTED] > 3.0db of gain is roughly equal to doubling the volume to the > human ear. Actually, that's not true. Each increase of 3.0 dB doubles the *power*. But the human ear's response is logarithmic, and the decibel scale is also logarithmic to reflect that. 1 dB is the smallest change in volume most people can perceive. ___ 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] Access Voicemail From Outside
> -Original Message-
> From: Bill Ford [mailto:[EMAIL PROTECTED]
> I'd like to see what some of you are doing to reliably aess
> voicemail from an outside line.
We have two ways of doing it.
One is a special extension from our IVR main menu. From the menu, users can
dial # followed by their extension number to access voicemail. This is a
carry-over from how our old system worked.
exten => _#XXX,1,MailboxExists(${EXTEN:1})
exten => _#XXX,2,SayDigits(${EXTEN:1})
exten => _#XXX,3,Playback(T-is-not-available)
exten => _#XXX,4,Goto(main-menu,s,1)
exten => _#XXX,102,Voicemailmain(${EXTEN:1})
exten => _#XXX,103,Hangup()
However, during the day the receptionist answers, instead of the IVR. To
allow people to still directly check their mail, we set up a POTS line as a
"back door number" into the system. The line is shared with an
outgoing-only computer fax modem, so the net cost of adding this was very
low.
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RE: [Asterisk-Users] TE405P and Dell Poweredge 6450 Incompatible?
> -Original Message- > From: Scott Stingel [mailto:[EMAIL PROTECTED] > I think there are TE410P compatibility issues with other > motherboards as > well. Google the archives (site:digium.com) under "HP > Proliant G4" for > example, as I remember some problems there. > > This response from Digium tech support, if quoted accurately, is not > acceptable to customers for obvious reasons. If there is a known > compatibility issue with some chipsets, I think it would be good idea > for Digium to publish them. Otherwise IMHO, they should > offer to accept > your return of the TE410P so you can pursue an alternative. What puzzles me is I don't know of any other adapter card that has these sorts of problems. It makes me suspect that Digium's design is a little marginal, so minor variations in motherboard design throw it off. FWIW, I have a PowerEdge 800 with two TDM cards in it. They work. Can't vouch for whether a TE405P would work in it, though. ___ 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] Voicemailbox detection:
See the "MailboxExists" application. If the mailbox exists, it branches to context n+101. -Original Message-From: Tim Connolly [mailto:[EMAIL PROTECTED]Sent: Monday, April 04, 2005 10:26 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Voicemailbox detection: Is there any way to detect if a user has a mailbox? I want to send all call which match _14XXX to voicemail except if the user doesn't have a voicemail box... ___ 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] vmail.cgi - can't forward messages
I'm having trouble with the 'Forward To' button in vmail.cgi. If someone tries to forward a message from the index screen, it logs this error: vmail.cgi: Invalid old Message If they try from the message playback screen, they get: vmail.cgi: Insecure dependency in mkdir while running setuid at /var/www/cgi-bin/vmail.cgi line 791. Anyone have a fix for this? I assume it's a taint-checking issue, but I haven't dug into it much yet. --- David Brodbeck, System Administrator InterClean Equipment, Inc. 3939 Bestech Drive Suite B Ypsilanti, MI 48197 (734) 975-2967 x221 (734) 975-4756 (fax) ___ 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
> -Original Message- > From: Scott Nelson [mailto:[EMAIL PROTECTED] > Perhaps you have an earlier hardware revision than I do; I also have > never rebooted the system. I have two TDM04Bs. If so, they must have sold me old stock. I bought the cards less than two months ago. ___ 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
> -Original Message- > From: Dana Olson [mailto:[EMAIL PROTECTED] > Alright, that helps clarify it a bit, but then again, I have been > running Asterisk at home with a TDM card for a couple months and > haven't had to restart it for a long time. Is it a requirement or just > simply a recomendation? If you haven't had any problems, there's no need to restart. The problem I have is that some modules on the card will simply stop responding. For example, the line connected to an FXO module will ring and ring, without it ever noticing. That's when a restart is necessary. In my case, it seems to happen after a couple of weeks of uptime, so I restart the card once a week on Sunday morning. ___ 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
> -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. It's often tempting to blame these failures on the design of other equipment (e.g., motherboards), or on power quality, or on things like that, but those are really just excuses for not providing a more robust product. Whenever I see a product that needs a lengthy "compatibility list" of motherboards that will and will not work with it, I get suspicious that someone is pushing the specs a bit. A simple example of this kind of thing that I'm familiar with is RS-232. As a hobbyist, I've put together various RS-232 interfaces. The RS-232 spec is something like -5 to -15V for a 1, and +5 to +15V for a 0. If you wire up a circult that uses 0V for a 1, and +5V for a 0, you eliminate the need for two power supply voltages, and it'll work great -- with about half the serial ports out there. On the other half, it'll fail miserably. ___ 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
> -Original Message- > From: Rich Adamson [mailto:[EMAIL PROTECTED] > I'll jump in here (but I'm not the original poster). The "once a week" > thing relates to the digium TDM card (fxo and/or fxs modules). I don't > believe the T1 cards are an issue that requires driver reloads. I'm the original poster. I probably should have qualified my comments. I have only had experience with the TDM cards. The T1 cards may be a different animal. ___ 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
> -Original Message- > From: Zoa [mailto:[EMAIL PROTECTED] > I didnt have to do a single restart in about 2 million calls on te4xpp > so far. I'm happy for you. But my TDM04B will stop responding after about two weeks, even if there are zero calls during that time. I found that out when I set up my test server. Currently, I have a script that shuts down Asterisk once a week, unloads the modules, then reloads everything. That keeps things in line, but it's a band-aid, not a fix. Currently, if I were looking to buy any sort of phone card, TDM or otherwise, Digium would not be my first choice. I also could not, in good conscience, recommend their products to other people. ___ 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
> -Original Message- > From: Brian Capouch [mailto:[EMAIL PROTECTED] > My understanding is that to an extent when we buy Sangoma > we're putting the dagger to Digium. If anything "puts the dagger" to Digium it'll be their own inability to engineer reliable hardware. I appreciate what Digium has done for Asterisk, but reliability expectations for phone equipment are extremely high. I sympathize with people who need hardware that doesn't need to be restarted once a week just to do its job properly. If Digium can't deliver on those reliability expectations, and do it soon, people are going to switch to companies that can. And you know what? I don't blame them. ___ 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] Dell 1750 & TDM400P - Power
> -Original Message- > From: Matt Schulte [mailto:[EMAIL PROTECTED] > I thought the TDM was "broke" on 1750's...?? I could never get passed > that NMI issue. I don't know about the 1750s. On my 800, loading the TDM modules the first time causes an NMI, but it seems to be harmless. Wish I could make that front panel light stop blinking, though. ;) ___ 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] Setting MWI on legacy PBX
> - Original Message - > From: "Brian S. Adelson" <[EMAIL PROTECTED]> > > You could probably utilize vmnotify to do exactly what you > are looking > > for: > > > > http://mikecathey.com/code/vmnotify/ Thanks. I may use that as a starting point if my home-grown solution doesn't work. I have a shell script that seems to be doing an okay job so far, though I won't know for sure until the system is in full use. ___ 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] Setting MWI on legacy PBX
> -Original Message- > From: David Brodbeck [mailto:[EMAIL PROTECTED] > How did you handle > the timing? I tried just throwing all the call files in at once, but > Asterisk doesn't take kindly to that. Just now I had the sudden inspiration to wait until there are no .call files before creating a new one. That seems to have fixed the problem. Don't know why it didn't occur to me sooner...it seems obvious, now. ___ 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] X100P voicemail volume too low (quiet)
> > I googled this a bit, and I saw similar complaints with > > older versions, but no resolutions. Also, many complained > > that it was only too quiet via email, but checking the > > message via SIP was OK. This isn't the case for me. It's > > just too quiet all around. On my TDM400 FXO cards, leaving and checking voice mail via the legacy PBX (very short loop length, in other words) is fine. On the POTS line I use as a voice mail back door, I had to boost the RX gain a lot (10 dB!) to get the voice mails and other recordings to end up the same volume as the pre-recorded prompts. (On the PBX lines I'm also cutting the TX gain by 6 dB, because it was deafening compared to outside phone calls otherwise. It's possible our Toshiba PBX just has very poor gain balance.) In either case, voice mails sent as email attachments are *very* quiet. I wonder if anyone has tried hacking in a script to normalize them before sending them? ___ 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] Setting MWI on legacy PBX
> -Original Message- > From: Henry Devito [mailto:[EMAIL PROTECTED] > By the #63 and #64 code it looks > like you are talking about a Toshiba PBX. At one time I actually wrote a > cron script that would check to see if there were messages in folders and if > there were it would generate a .call file that would dial the #63+ext. If > there were no messages it would dial a #64. I dedicated 1 port to do this > for approximately 40 phones. Yeah, that sounds like pretty much exactly what I need. How did you handle the timing? I tried just throwing all the call files in at once, but Asterisk doesn't take kindly to that. It tries to dial them all at once and I get "app_queue.c:374 in changethread: Can't change device '**Unknown**' with no technology!" errors. ___ 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] Setting MWI on legacy PBX
Before I go and try to write something myself, I'm curious if anyone has a script that they're using for setting and clearing the MWI on a legacy PBX. I need to pick up a Zap channel and dial #63XXX to set the MWI, or #64XXX to clear it, where XXX is the extension number. One complication is that I've got a couple dozen extensions to handle the MWI for, and only four channels to work with, so I'll need to either only set or clear the ones that have changed, or queue up the callouts somehow. --- David Brodbeck, System Administrator InterClean Equipment, Inc. 3939 Bestech Drive Suite B Ypsilanti, MI 48197 (734) 975-2967 x221 (734) 975-4756 (fax) ___ 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] IRQ headaches
> -Original Message- > From: Brett, Gary [mailto:[EMAIL PROTECTED] > I have moved off of IRQ 10 and onto IRQ 5, but everytime I > boot up, I get > usb-uhci and ehci_hcd using IRQ 10 as well as my Digium card. > Does anybody > know what these are and how I can get rid of them ? They're USB ports. If you don't need USB, disable the USB controller in the BIOS and they'll go away. ___ 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] Why isasterisk's voice mail calledcomedian.
> -Original Message-
> From: Mark Charlton [mailto:[EMAIL PROTECTED]
> Plus if you send your users to VoicemailMain(${CALLERIDNUM})
> they don't hear
> it at all.
> They just get "enter password".
Yup. If you do that, the only time they hear it is during the initial setup
call (if you have "forcename=yes" or "forcegreetings=yes" set.)
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RE: [Asterisk-Users] What causes this changethread error message?
> -Original Message- > From: David Brodbeck [mailto:[EMAIL PROTECTED] > I'm running Asterisk HEAD from March 4. I've Googled a bit > but I can't > figure out what causes this error: > > app_queue.c:374 in changethread: Can't change device > '**Unknown**' with no > technology! > > It doesn't seem to be causing any problems, but I'm curious > what causes it. Never mind. It was from something I was doing with an external script. ___ 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] What causes this changethread error message?
I'm running Asterisk HEAD from March 4. I've Googled a bit but I can't figure out what causes this error: app_queue.c:374 in changethread: Can't change device '**Unknown**' with no technology! It doesn't seem to be causing any problems, but I'm curious what causes it. I did a few Google searches and found a lot of people asking about it, but no real answers. ___ 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 Capabilities
Title: Asterisk Capabilities -Original Message-From: Parker, Blake (MIS) [mailto:[EMAIL PROTECTED]Sent: Wednesday, March 16, 2005 12:41 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk Capabilities I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? [David Brodbeck] Yes. Look into "dialplans" and "extension contexts". ___ 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: Best DB
> -Original Message- > From: Giudice, Salvatore [mailto:[EMAIL PROTECTED] > As for your 'artist license with your data' comment, put it into some > context. I would blame a programmer for trying to insert a > string of 255 > characters into a field only 100 character wide. Maybe you could blame > the dba for not building a schema to support the application. > Regardless, I would not call the database deficient because > it truncates > your data to 100 characters and doesn't warn you with an error. And the sad fact is, if the software isn't doing any data verification, it's probably not doing error checking either. So if the DB throws an error, your database will be protected, but the application will probably crash or do something undefined. Which of those situations (truncated data, or a crashed app) is better depends on the application. It's not clear cut. ___ 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: Best DB
This Postgres vs. MySQL business is ultimately just a religious debate, like PC vs. Mac, Ford vs. Chevy, or Kirk vs. Picard. They both work; they both have their plusses and minuses; and debates about which are better never convince anyone to change their preconceived ideas. It's also about as on-topic for this list as any of the other subjects I just mentioned. ___ 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: Best DB
> -Original Message- > From: Steven Critchfield [mailto:[EMAIL PROTECTED] > > Top Deployed Databases poll shows following databases in use: > > > > SQL Server with 78%, Oracle - 55%, MySQL - 33% and PostgreSQL - 8%. > > I see they created this with Mysql, > 78 + 55 + 44 + 8 = 185% > I'm sure if you add in the others we would get to something > around 300% > deployment. Presumably some sites had more than one type of database in use. ___ 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: Best DB
I could start a pretty big flame war if I tried to compare Windows 95 with MacOS X by deployment stats instead of stability. [David Brodbeck] I've seen Mac OS X locked up solid just by putting in a damaged CD-R disc. It's a nice OS, mind you, but it's not as stable as some people would lead you to believe. ___ 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] Voicemail SMS Alert - Possible?
> -Original Message- > From: Julius Kidubuka [mailto:[EMAIL PROTECTED] > I need to be able to send an sms alert to one's mobile/cell phone. For > instance, when I receive a voicemail message in my inbox, I > also want to be able to get a message on my cell phone alerting me of this > e-mail. How possible is this? And if it is, what do I need to do to get > the service up and running? > > Ideas are most welcome. If your cell phone service offers an email-to-sms gateway, putting the email address in the pager_email field of voicemail.conf works pretty well. That may be more of a U.S. thing, though. ___ 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 lock up
As far as I can tell, the main kernel configuration issue is to make sure APIC and IO-APIC support is turned on, if your system supports it. (All SMP motherboards do, and many single-processor ones do as well.) This seems to give access to 32 interrupts instead of 16, minimizing interrupt sharing. -Original Message-From: Dennis Webb [mailto:[EMAIL PROTECTED]Sent: Friday, March 11, 2005 2:51 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] TDM04B lock up Should ACPI be turned off in the kernel? In the bios I can only set cards to the 1-15 interrupt range, but linux and acpi it seems moves these to the 20's. I looked last night on the lists and found no true answer to this question. I have 4 TDM's so interrupts below 15 are few and far between, but I do have enough. ___ 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] Play music on hold while waiting for DTMF?
Is there a way to play music on hold for a specified amount of time while listening for DTMF? I suppose I'm looking for a hybrid of Background() and WaitMusicOnHold(). I don't really want to use Background() because the music would start over each time. ___ 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: [OT] - [Asterisk-Users] Why should I answer a Newbie questio n,therethick!
> -Original Message- > From: Paul Fielding [mailto:[EMAIL PROTECTED] > Frankly, I agree. If you don't like the question, feel it's > lame or dumb, > or don't like that someone hasn't done their research, then > delete the message. Well, sometimes that works. But I've been on a lot of lists where newbies who thought they were being ignored started flaming people for not responding to them, writing posts badmouthing the project, hijacking other threads, accusing people of being cliquish, etc. Sometimes you just can't win. ___ 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: [OT] - [Asterisk-Users] Why should I answer a Newbie questio n, therethick!
> -Original Message- > From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] > Sometimes it is not the "if" you make a search, often is for > new comers > "what" to aks for. > If you do not know the specific term, than you need to ask somewhere, > and I think the list is good for that. Sure. So say, "I tried a Googling for X, but I didn't have any luck. Then I looked at pages X and Y in the Wiki, but couldn't find anything that related to my problem." People are a lot more sympathetic if you demonstrate you've made some effort to find the answer on your own. ___ 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] kernel error with Zaptel cards
-Original Message-From: Christopher [mailto:[EMAIL PROTECTED]Subject: Re: [Asterisk-Users] kernel error with Zaptel cards I just would like to not have that damn status light flashing all the time. It hard to explain to people who walk in the server room :) I know what you mean. I just say, "no, it's supposed to do that." ;) I wonder if there's any way to reset that light in software? Then you could just run a program to clear the error, after loading the wctdm modules. That's still a hack, though. ___ 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] kernel error with Zaptel cards
> -Original Message- > From: Christopher [mailto:[EMAIL PROTECTED] > I see that there is a lot of discussion on the web about a > common error > after installing and modprobe'ing the zaptel driver. However I don't > see any resolution, anyone found a solution? > > Here's the output I get after modprobe: > > Uhhuh. NMI received. Dazed and confused, but trying to continue > You probably have a hardware problem with your RAM chips > Module 0: Installed -- AUTO FXO (FCC mode) > Module 1: Installed -- AUTO FXO (FCC mode) > Module 2: Installed -- AUTO FXO (FCC mode) > Module 3: Installed -- AUTO FXO (FCC mode) I see this too, on a Dell PowerEdge 800. However, it doesn't seem to have any ill effects, other than making the status light on the front panel flash orange. I only see it the first time the modules are loaded. If I rmmod them, then load them again, I don't get another NMI. ___ 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] List tips for new subscribers
> -Original Message- > From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] > On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote: > > Oh I'm sorry. This is the first list I've joined where this > is such a big > > issue! Forgive me for not having your superior > understanding of mail > > clients, and/or list servers! > > You have a *servere* inferiority complex. > > I asked a simple question. The only people who don't see why > it's a problem > use inferior mail user agents which don't support threading, > or perhaps they > don't realize that they can do threading. FWIW, I was unaware of this issue for a long time, too. I did use threading in Outlook, but like many clients it "fakes" it using the subject headers. It wasn't until I started using Thunderbird elsewhere that I understood what people were complaining about. ___ 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] compiling cvs-head today?
> -Original Message- > From: Steven Critchfield [mailto:[EMAIL PROTECTED] > Looks like a hardware problem as you had failures in > different locations > but both where a gcc seg fault. This means either your CPU is hot and > starting to spit out randomness or your memory is failing and > producing > randomness. Could be something else like low power supply and therefor > faulty writing/reading of data to/from memory. > > Any way around it looks like you are in for either a while of > debugging > hardware or a hardware replacement regiment. The first thing I usually do in these situations (after making sure the machine's fans are all running and dust-free) is run MEMTEST-86. http://www.memtest86.com/ It's not foolproof, but in my experience it catches more memory problems than any other utility. ___ 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: How do I match a "D"? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
> -Original Message- > From: Chris Wade [mailto:[EMAIL PROTECTED] > Brian Roy wrote: > > I think that my PBX does this too. Is there any way I can get the > > Zaptel drivers to disconnect on that tone too? I would love > to replace > > my existing voicemail with * but I can't get my PBX to signal a > > disconnect properly. I have to use busycount=10 but every voicemail > > has an annoying busy signal tacked onto the end of it. > > CVS HEAD has a features.conf entry for 'disconnect' which is normally > set to '*', you might try placing a 'D' there and see if that works? I just played with that a little, but it doesn't seem to do anything as far as voicemail is concerned. Maybe I'm missing something. It'd be really nice if there were a way to get voicemail (and other apps, like Directory()) to properly interpret this. My phone system doesn't give a busy after a hangup, just a 'D' followed by silence, so right now I'm coping by setting "maxsilence=5" and "review=no" in my voicemail.conf. Not ideal, but it works. I've considered trying to hack the voicemail source code to handle the 'D', but I haven't really dug into it. ___ 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: How do I match a "D"? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
> -Original Message- > From: Paul Rodan [mailto:[EMAIL PROTECTED] > I'll ask a stupid question, how does a user hit an alpha > letter from his touchtone? > > I know that the Cisco 7960's support entering alpha letters, > and it could > potentially do it (maybe), but how does the average end user > enter an a b c or d from their touchtone phone? They don't. Most phones lack the fourth column that has those keys. Some PBX phones have it, though, and it's common on 2-way radios that include DTMF keypads. (Believe it or not, before the advent of cell phones some businesses provided a "phone patch" interface to allow making phone calls from their 2-way radios. This is still quite common in the railroad industry, AFAIK.) In my case, I need it because it's how my PBX does disconnect notification to the voice mail system. When the line is hung up, it sends a "D". I expect these digits are very rarely used, which is probably why no one thought to document that Background() ignores them. ___ 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] Background() ignoring digits A-D (Was: RE: How do I match a "D"?)
> -Original Message- > From: Steven Critchfield [mailto:[EMAIL PROTECTED] > On Wed, 2005-02-09 at 09:15 -0700, Kevin P. Fleming wrote: > > David Brodbeck wrote: > > > > > Anyway, I figured it out. The extension was working, but > Background() > > > ignores the tones A through D by default. I didn't > realize this because I > > > wasn't waiting for message playback to finish. > > > > Please enter a bug in Mantis for this; it should very likely be > > corrected, as I don't see any reason to ignore A-D in Background(). > > I would suggest that that be added as a optional switch to > background to > get the extra digits. While I do not know if A-D are easy to hit with > Talk-Off, it is 4 more potential digits to hit. Also it would be less > surprise to a user to be required a flag to Background() to get those > than it would be to diagnose why you occasionally get dumped in an > invalid extension. At very least it should be documented. It was very confusing to see the digits come up in the debug output without provoking any response from *. Ideally, I suppose, Background() would ignore *any* invalid digits, but that would require it to understand the dialplan, which is probably impractical. It would minimize the chances of talk-off, though. (Why are people talking during the menus, anyhow? ;) ) ___ 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: How do I match a "D"? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
> -Original Message- > From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] > David Brodbeck wrote: > > > Anyway, I figured it out. The extension was working, but > Background() > > ignores the tones A through D by default. I didn't realize > this because I > > wasn't waiting for message playback to finish. > > Please enter a bug in Mantis for this; it should very likely be > corrected, as I don't see any reason to ignore A-D in Background(). http://bugs.digium.com/bug_view_page.php?bug_id=0003538 A simple fix is included, though I don't have a deep understanding of the Asterisk code, so it's possible it has side effects I'm not aware of. ___ 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: How do I match a "D"? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
> -Original Message- > From: Gilad Ben-Yossef [mailto:[EMAIL PROTECTED] > I'm prbably stupid, but wont this do what you want? > > > exten => 1,1,Goto(bye,s,1) No, because I wanted to match on "D", not "1". Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. ___ 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] Bug? Background() doesn't recognize D tone.
> -Original Message- > From: David Brodbeck [mailto:[EMAIL PROTECTED] > I finally figured out my extension D issue. The extension > works fine as > long as Background() has finished playing. But during > playback, the "D" > tone is not recognized. Is there any way to configure this? > Is this a bug? Sorry to keep responding to myself, but I figured it out. Background uses AST_DIGIT_ANY as a list of digits to look for. In /usr/src/asterisk/include/asterisk/file.h, we find this line: #define AST_DIGIT_ANY "0123456789#*" Changing it to #define AST_DIGIT_ANY "0123456789#*ABCD" gave me the behavior I expected. I suppose, technically, A, B, C, and D are not digits...but then, neither are # and *. ;) ___ 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] Can someone tell me why I'm getting these? ( mailing list probe message)
> -Original Message- > From: Andrew Thompson [mailto:[EMAIL PROTECTED] > Twice in the last week or so, I've received a message similar to the > attached. > > A portion of the attachment that's attached is not in > English. Is this > my mail server failing, or someones who's on the list? I got one of these, too. It looks like the message I sent to the list bounced when the list sent it to someone else, and the list software interpreted it as a bounce from *me*. Something is pretty broken. ___ 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] Bug? Background() doesn't recognize D tone.
I finally figured out my extension D issue. The extension works fine as long as Background() has finished playing. But during playback, the "D" tone is not recognized. Is there any way to configure this? Is this a bug? ___ 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 do I match a "D"? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
> -Original Message- > From: David Brodbeck [mailto:[EMAIL PROTECTED] > Okay, the problem appears to be that I'm tone deaf. ;) > > I finally thought to turn on debugging on the channel. The > PBX is sending > "D", not "*". The programmer of the previous voice mail system (whose > configuration I was cribbing from) seems to have made the > same mistake. Is there some trick for matching the "letter" tones? I added this extension: exten => D,1,Goto(bye,s,1) But it doesn't trigger, even though I see this debugging output when I hang up: << [ TYPE: DTMF (1) SUBCLASS: D (68) ] [Zap/1-1] ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
Okay, the problem appears to be that I'm tone deaf. ;) I finally thought to turn on debugging on the channel. The PBX is sending "D", not "*". The programmer of the previous voice mail system (whose configuration I was cribbing from) seems to have made the same mistake. ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
> -Original Message- > From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] > On February 8, 2005 09:28 am, David Brodbeck wrote: > > What puzzles me is it works fine if I dial *, but if I hang > up instead and > > the PBX sends *, Asterisk doesn't seem to get it. > > With you listening in on the same physical 2-wire that the > PBX uses and you > send *, does Asterisk see it? If you're on a call from the > PBX to Asterisk and dial * from the PBX phone, does * see it? Yes, in both cases. ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
> -Original Message- > From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] > On February 8, 2005 08:44 am, David Brodbeck wrote: > > The sequence I hear on the extension, when I plug in an > analog phone, is > > the click of the phone at the other end being hung up, > followed immediately > > by a * touchtone. Then there's silence until I hang up. > > Hmm... I bet it has everything to do with not having 't' or > 'T' in the > dialplan -- asterisk is ignoring the tones because it's potentially a > security problem. I have a "t" in the dialplan, but not "T". I could add a "T" entry and see if it'll help. Currently what happens is Asterisk doesn't seem to notice the *, then it eventually goes to the "t" extension. This is undesirable since "t" transfers to the receptionist, who then gets a dead call. What puzzles me is it works fine if I dial *, but if I hang up instead and the PBX sends *, Asterisk doesn't seem to get it. > At least that's my current working theory -- I am not sure if > t/T listen ofr > all DTMF or just #, and I also don't know the direction of > your call (PBX -> * or * -> PBX). PBX -> Asterisk. I'm setting up Asterisk to replace an old voice mail system. ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
> -Original Message- > From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the phone at the other end being hung up, followed immediately by a * touchtone. Then there's silence until I hang up. ___ 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] In-band disconnect problem (legacy PBX) - asterisk doesn't hear t he touchtone?
The legacy PBX I'm working with does in-band disconnect notification -- it sends a * touchtone when the line is hung up. I've been trying to get this to work with Asterisk. I added a * extension to my menu context that plays "Goodbye" and hangs up. This works fine if I manually press *, but it never triggers when I hang up and the PBX sends it. I've plugged in an analog phone on a splitter and verified that the tone is being sent after I hang up. Any ideas why Asterisk isn't hearing it? --- David Brodbeck, System Administrator InterClean Equipment, Inc. 3939 Bestech Drive Suite B Ypsilanti, MI 48197 (734) 975-2967 x221 (734) 975-4756 (fax) ___ 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?
> -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > 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? :) A phone number is more analogous to an IP address than a domain name. If you move, you'll have a different ISP, and you won't get to keep your old IP address. Your domain name, however, can be pointed to any IP address you like. It's that extra layer of indirection -- the domain being resolved to an IP -- that lets Internet domains be moved so easily. It's also partly a historical issue. The phone system is layed out geographically, because in the days of mechanical switches that was the only reasonable way to do it. Each area code represents a certain area of the country, and each exchange (the first three digits of the local number) represents a particular central office. If you're outside the area covered by that central office, there's no way to get a direct line run to you (unless you use forwarding, or something like VOIP.) Billing is based on this, too. If people could move numbers around willy-nilly, you'd never know if you were making a long-distance call or not. ___ 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] Re: load balancing 20 asterisk servers
> -Original Message- > From: Rich Adamson [mailto:[EMAIL PROTECTED] > You have fiber-seeking-backhoes in your area? Wow! They're everywhere, man! When I was in college an entire nearby town lost all phone service for 24 hours due to a backhoe cutting a fiber optic cable. 3,000 people with no way of calling emergency services for an entire day. ___ 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] Outlook Integration
> -Original Message- > From: Manjit Riat [mailto:[EMAIL PROTECTED] > The partner list shows digium as one of their partners. So > under GPL they > have to provide the source code for the app. Not unless they're linking with a GPL library, or using source code from a GPL app. If it's only accessing Asterisk through the network I don't see how the GPL's viral clause applies. Otherwise every web browser that accessed an Apache server would have to be GPL'd. ;) ___ 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] TDM400 stopped working
> -Original Message- > From: Matthew Laird [mailto:[EMAIL PROTECTED] > Hmm, found the problem, I just manually ran it again (I did > last night) > specifying the configuration file well that's annoying. I have > zaptel.conf in /etc/asterisk along with the other configs, ztcfg looks > in /etc So, why does it expect the file somewhere else > from all the other asterisk configuration files? Left hand doesn't know what the right hand is doing, I'd guess. The people who wrote the hardware driver and the people who wrote Asterisk may not be the same. ___ 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] TDM400 stopped working
> -Original Message- > From: Matthew Laird [mailto:[EMAIL PROTECTED] > I excitedly installed my TDM dev kit earlier this weekend, installing > asterisk and all the kernel drivers to make it work. And it > did, it was > fantastic. > > I then reboot the machine, and even after doing a modprobe > wctdm, I get > the following: > == Parsing '/etc/asterisk/zapata.conf': Found > Jan 31 13:34:27 WARNING[342]: chan_zap.c:793 zt_open: Unable > to specify > channel 1: No such device or address Did you remember to run ztcfg after loading the module? You have to do it every time or the channels won't be configured. ___ 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] Re: UPS for Asterisk
> -Original Message- > From: Michael Loftis [mailto:[EMAIL PROTECTED] > Not old, just small it seems. The little Norstar (merlin?) > Nortel's do > NVRAM/Flash, as do Panasonic's. There's also the App/VM > Module which is an > OS/2 based system, or was. Toshiba Strata systems also use NVRAM to hold their configuration. They pitch this as a selling point -- no moving parts. I think their integrated voicemail system uses a hard disk, though. ___ 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] UPS for Asterisk
> -Original Message- > From: Michael 'Moose' Dinn [mailto:[EMAIL PROTECTED] > You should be able to boot Asterisk using slackware as a base > from a 64M CF > card or even from a 64M bootable USB memory key. If you use > ReiserFS or > something similar for the drive that stores all your > voicemail, etc then it > should come back without a problem as well. > > > Of course you want to make sure the system shuts down cleanly too... Or use a journalling filesystem. ___ 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] TDM400 in aging Dell Optiplex
> -Original Message- > From: Jeff Pratt [mailto:[EMAIL PROTECTED] > Got one running in an Optiplex GX100. Works fine. I put one in an Optiplex GX170, for testing. In the topmost PCI slot, it couldn't generate interrupts. Worked in the next slot down, though. TDM400 + Dell seems to be a finicky combination. ___ 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] UPS for Asterisk
> -Original Message- > From: Shoval Tomer [mailto:[EMAIL PROTECTED] > That's not a problem. > The question is what happens when the power's restored. > > Can you go ahead and just start working or do you need to call the > technicians to come reconfigure the whole thing? It comes back up on its own, of course. > If it just works, you have something asterisk without UPS can't give > you. Really? Surely Asterisk can be configured to start itself up when the system boots. ___ 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] UPS for Asterisk
> -Original Message- > From: Peter Svensson [mailto:[EMAIL PROTECTED] > The SmartUPS ups's from APC that are >= 1kVA seem to be of a > lot better > quality then their smaller siblings. We have lost none of the 1kVA or > larger ups:es while several of the smaller ones have died of > electronics > failures. The larger ups's seem to have 5-year batteries > while the smaller > ones have 3-year batteries. Just a counterexample -- the batteries in our 3 kW rack-mount SmartUPS only lasted three years before one of them failed. They were a real pain to get out, too -- the failed battery had swelled up and wedged itself in the battery compartment. The rack-mount UPS runs around 120 degrees F inside and that seems to shorten the battery life. ___ 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] Menu tree for voicemailmain application
> -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > On Mon, Jan 24, 2005 at 10:55:03AM -0500, David Brodbeck wrote: > > Is there a menu tree diagram somewhere for the > Voicemailmain application? I > > know my users will ask for one, and before I started > drawing my own I > > thought I'd see if someone already had. > > Ask the wiki for voicemailmain, use the fourth link, done. Thanks. I feel dumb for not spotting that, now. :) ___ 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] UPS for Asterisk
> -Original Message- > From: Jon Radon [mailto:[EMAIL PROTECTED] > I've had good luck with CyberPower, what was your issue? I had two of them. The first one, after about a year, would just randomly switch off or glitch, causing the computer connected to it to reboot. The second one lasted two or three years, then suddenly started acting like the incoming power was off, even when it wasn't. It did this briefly, intermittently for a couple of months, and then the condition became permanent and it would no longer switch to the AC line or charge its batteries. I gave up on the brand at that point, figuring an unreliable UPS was even worse than none at all. ___ 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] Menu tree for voicemailmain application
Is there a menu tree diagram somewhere for the Voicemailmain application? I know my users will ask for one, and before I started drawing my own I thought I'd see if someone already had. --- David Brodbeck, System Administrator InterClean Equipment, Inc. 3939 Bestech Drive Suite B Ypsilanti, MI 48197 (734) 975-2967 x221 (734) 975-1646 (fax) ___ 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] Best VPN server for * and woad warriors usin g windows?
> -Original Message- > From: Remco Barende [mailto:[EMAIL PROTECTED] > Why is it bad to put a vpn server on the * box? CPU load. IPSec can be quite CPU intensive. So can asterisk. Putting two CPU-intensive, time-sensitive applications on one machine is asking for trouble. It may work, though, if you don't have too many simultaneous users. > I was considering IPSEC because I heard it is safer or more > secure than PPTP and Windows XP supports IPSEC natively (or so it > claims). It does, but I've never had much luck getting it to interoperate with anything but a Windows server. I've heard it can be done, but I don't understand IPSec well enough to make it work. It's not simple. ___ 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] UPS for Asterisk
> -Original Message- > From: Jon Radon [mailto:[EMAIL PROTECTED] > Why risk it? Just go snag a cheap UPS from your local store. Just > get something with enough run time to shut the system down gracefully. Don't go *too* cheap, though. I had a couple of really cheap (under $40) CyberPower UPS's that ended up causing more outages than they protected against. I've had good luck with APC, but keep in mind that the batteries have a finite lifespan. On SmartUPS and BackUPS Pro models, you'll get a warning that the battery needs replacing, but on regular BackUPS models the first hint you get that the battery is bad is when the power goes out and the UPS doesn't work. This is sometimes okay for workstation use, but I'd hesitate to put one of those on a server. I find that the batteries in our APC UPS's generally last four to five years for stand-alone units, three years for rack-mount ones. ___ 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] UPS for Asterisk
> -Original Message- > From: Shoval Tomer [mailto:[EMAIL PROTECTED] > On the other hand, telephony down time is unacceptable. PBXs have a > counter part. Plain old PBXs are expected to run 24x7. real 24x7, with > uptimes of 99.999. And if you think about it, they actually do. That would be news to the people who installed our (non-Asterisk) PBX. It has no battery backup at all. When the power goes out, so do all our phones. (Except for the fax machines, which don't go through the PBX.) ___ 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] Newbie question - can't get Asterisk to pick up incoming call
Got it working. The card was unable to generate interrupts in that PCI slot, but I moved it to another one and it works now. Thanks for your help, Roger. ___ 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] Newbie question - can't get Asterisk to pick up incoming call
Okay, I'm going to preface this by saying I'm sure I've overlooked something really basic here. I just need someone to hit me with a clue stick and point out what I'm missing. I've got a TDM card with four FXO modules. I've plugged one of them into a PSTN line. I'm working through the examples in the Asterisk Documentation Project guide, but I can't get Asterisk to answer the line. I don't get any diagnostics on the Asterisk console when the line rings (should I?) Here are my config files: /etc/zaptel.conf: fxsls=1-4 loadzone=us defaultzone=us /etc/asterisk/zapata.conf: [channels] language=en context=default switchtype=national signalling=fxs_ls channel => 1-4 /etc/asterisk/extensions.conf: [default] exten => s,1,Answer() exten => s,2,Playback(goodbye) exten => s,3,Hangup() The zaptel and wctdm modules are loaded. ztcfg -vv gives this output: Zaptel Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Slaves: 01) Channel 02: FXS Loopstart (Default) (Slaves: 02) Channel 03: FXS Loopstart (Default) (Slaves: 03) Channel 04: FXS Loopstart (Default) (Slaves: 04) 4 channels configured. What am I missing, here? ___ 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] long delays in list posts?
> -Original Message- > From: Matthew Boehm [mailto:[EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] long delays in list posts? > OMG! 1 hour?!?! I just now got this at 4:40PM. It takes an hour for my > emails to get posted to the list? Geez.. The fact that it's almost exactly one hour suggests to me that maybe they're doing teergrubbing. ___ 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] Looking for a wireless phone... wifiortradit ionalwireless ?
> -Original Message- > From: Steve Totaro [mailto:[EMAIL PROTECTED] > Here is the DC metro area I drove to work with my laptop > running network > stumber. In the 35 miles it takes to get to the office I stumbled 310 > wireless networks and more than half of those were wide open. > At least in this area, a wifi phone can almost replace a cell. Only if you hold still, I would think. Wouldn't the call drop every time you handed off to a new network and got a new IP address? ___ 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] Connecting Asterisk to a Toshiba Strata syst em
> -Original Message- > From: Brian Roy [mailto:[EMAIL PROTECTED] > I do this with a Toshiba Strata DK280. The model before the CTX. You > shouldn't have any problem doing this with the CTX either. Assuming > you don't have SMDI integration with your voicemail (you didn't state > that you did), you just need the phones to send trailing digits, and > you need to negotiate the lighting of voicemail lights on phones. All > manageble from Asterisk. Nope, no SMDI integration. I'm pretty sure that the trailing digits are being sent already, for the current CallXpress voicemail system. I guess mostly I'm just not certain about how to set up Asterisk to handle this sort of configuration. I've been looking through the documentation, but it all seems to assume that I want to set up a full stand-alone PBX. Thanks for your suggestion to plan for more ports than I currently need. I can certainly see the wisdom in that! ___ 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
