Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
Vicky [EMAIL PROTECTED] wrote: I have 3 toll free did's with nufone since 1 month .. Until now i dont have a problem with them .. their portal was good enough to do proper configuration and call quality wasnt bad ( even though i havent used them in really huge traffic yet ) . The reason for the warning is to get people to move before they start to have problems. I'm sure everything will work for a while (perhaps for years), but when you have a problem - perhaps an sudden deterioration or complete outage at NuFone itself, you'll find out just how difficult it is to get a helpful, or even just polite, response from their support department. You might find that you have no service through no fault of your own, and have no way to fix it yourself. Usually a threat of fix it or I'll take my business elsewhere tends to work, but if you do follow up on that then NuFone will rob you of any outstanding call credit you have in your account. You'll find lots of examples of this if you search around. I ignored the warnings because the service was working at the time, and I didn't think I'd ever need to contact their support department. Don't make the same mistake. Here are a few references I found with a five-minute Google search. There are lots more where these came from: https://forum.voxilla.com/other-providers/nufone-severe-problem-service-9548.html http://www.voip-info.org/wiki/view/Nufone http://www.shakataganai.com/index.php?/archives/163-NuFone.html http://hansgrueber.blogspot.com/2006/01/nufone-sucks.html There are loads of examples in the various mail list archives. Here are a couple: http://lists.digium.com/pipermail/asterisk-biz/2004-December/001468.html http://lists.digium.com/pipermail/asterisk-biz/2004-December/001457.html As you have only had the DDIs for a month, you could get away with simply going elsewhere and using new numbers (use up your call credit first). If you have vanity numbers, or you want to keep them for some other reason then I wish you good luck getting NuFone to help you port them to your new provider. The bottom line is that you are free to use whichever provider(s) you like, but please be aware of the high likelyhood of having severe service difficulties with NuFone. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
Andrew Joakimsen [EMAIL PROTECTED] wrote: NuFone isnt bad if you want a disposable termination account. But don't rely on it for anything. Well, the voice quality left a lot to be desired, so I didn't make a lot of use of the service anyway. Perhaps, if I had made more use of it, they wouldn't have been able to steal as much from me when I fired them for incompetence, laziness and general rudeness. Perhaps they think that if they rob enough people then they will be in a better position to pay their upstream provider - leading to less downtime and therefore less cause for customers to attempt to contact the idiot running their support department (Jeremy). They are in urgent need of a better business plan (one that doesn't rely upon raising money by stealing funds from customer accounts) and a complete change of staff. Oh, and a clue! -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
Doug Crompton [EMAIL PROTECTED] wrote: If this (or any) company is really stealing or not living up to a contract then why not report them as such, especially if they are US based. I would suspect you would have another route to take. I am UK-based - there's not a lot I can do to a US-based company, and they know it. That is unless I'm willing to fly over there to sue them, which I'm not. In this case, it was better to just cut our losses and make sure as many others as possible know about our experience and are forewarned. I have already convinced several others to abandon NuFone. I was the one who recommended them in the first place, so I felt that I had to warn them and try my best to persuade them to migrate. Remember - sometimes, when you annoy one customer, you loose a lot more than just that one customer. Whatever playground victory NuFone thinks they won by helping themselves to the content of our account, they lost hundreds of times over in terms of the future revenue from the customers I know that they lost. If you don't do anything about it then they will just go on abusing others and getting away with it. That's why I feel it is right and proper to warn others. If NuFone carries on operating they way they do, they will lose a lot more than just the customers we had powers of persuasion over. The snowball effect induced by mounting bad publicity is a powerful thing, and not something any company wants to be on the receiving end of. I know that I have caused more damage this way than I could ever do by simply recovering the account balance in court, and I didn't have to fly anywhere to do it. NuFone's short-sighted and clearly criminal ways will come back to haunt them one day. At the very least the BBB (bbb.org) should be notified. They have a web site and if it is really wire/internet fraud then the FBI (www.fbi.gov/majcases/fraud/internetschemes.htm) has a site you can register a complaint with. I've never heard of the BBB. I have now - thanks. I doubt that NuFone's behaviour counts as fraud though. I'd class their actions as just plain old-fashioned theft. If you are a NuFone customer then I advise you to use up your balance and leave as soon as possible. It's very easy to do - especially if you're only using a company as an outgoing route and don't need to port a number to a new provider. If you know any NuFone customers then you should try to get them to do the same. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
Doug Crompton [EMAIL PROTECTED] lazily top-posted: Well if this company is US based I would not think where you are matters if it is fraud. You could still enter a complaint at the FBI site. I would also think they would be working with the UK counterpart. I doubt it - not on a minor theft case, perpertrated by minor criminals. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
www.IPKall.com [EMAIL PROTECTED] wrote: I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Anyone except NuFone. Their customer service is non-existant - you have to email every day for a couple of months before you'll be privileged enough to get a one-line response to a service outage issue. If you dare to point out that the response didn't address the issue then you'll unleash the combined wrath of both of the brain cells in residence at NuFone's support department. Not immediately, of course - you'll have to wait another couple of months for a reply. If you give up on them and decide to go elsewhere, they will pocket any outstanding funds you have pre-paid into your account. Existing NuFone customers are advised to not pre-pay too much to these yokels, and to jump ship as soon as possible. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Largest working config files?
Steve Davies [EMAIL PROTECTED] wrote: I hope this is not a FAQ - I have not been able to find it if it is covered already... I have a dial-plan on my asterisk system that is becoming potentially quite large and complex - Of the order of 12 lines of dialplan per extension number. Most of this is in order to record suitable CDR data, access voicemail, and play polite goodbye messages etc. The operation of each extension can potentially be unique, making a common [extensions-generic] almost impossible to write. Have you looked into creating a couple of macros to reuse your code? [local-extensions] exten = 2100,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN}) exten = 2101,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN}) exten = 2102,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN}) exten = 2103,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN}) ... As you can see, I only need one line per extension; All of the call logic is in the [macro-call-local] macro. The maintenance is a lot simpler too, of course. Does anybody have experience of how big an extensions.conf can get before problems start occuring? If anyone has experienced problems, what sort of things happen? I have no idea. Here's our dialplan line count (quite small because of the macros): 218 extensions/incoming.conf 354 extensions/internal.conf 225 extensions/macros.conf 471 extensions/outgoing.conf 151 extensions/routes.conf 1419 total -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: www.openpbx.org
Brian C. Fertig [EMAIL PROTECTED] wrote: Further info. The domain is registered to Marc Olivier Chouinard. He has posted in the dev list. Can they do this? Is this legal? Yes - anyone can register a domain name. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: www.openpbx.org
Jon Pounder [EMAIL PROTECTED] wrote: There are people out there who wish to contribute, and not have their work lost on an individual project website since they do not choose to accept digium's terms to contribute to asterisk. This gives them an opportunity to do so, and have their work aggregated with everyone else in the same category, so it is one stop shopping for users. Open source is about choices, not restrictions, and this gives contributors more choice. As long as the two streams stay compatible (which they likely will) it should be better for everyone. I think it's a great idea, and long overdue. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
Matt Riddell [EMAIL PROTECTED] wrote: My comment was directed at the USPTO who grants patents on a regular basis with what seems like no effort to check for prior art. The first time I saw this I thought it was stupid. The second, unbelievable, and the 358456347563th one crazy! The life of a US patent clerk must be so boring. The greatest excitement they probably get is when they run out of ink for their approved rubber stamp. The USPO could probably outsource their approval process to a third-world sweat shop. If patent clerks were forced concentrate on their work, there would be no atomic bomb. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote: Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge. Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. Marvellous. Another company with a monopoly over aspects of VoIP technology. I don't have the millions required to mount a defence in a North American court, so I should just consider myself lucky that I live in a free country. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel Pentium Celeron
Giordano Grandis [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) i'm going to install asterisk with a 4 BRI (HFC chipset) on a Celeron at 2.6 GHz I dont known Celeron performance, but i listen that is not very good. Could I have some performance isuue with this kind of processor ? You could have performance issues with any processor; It all depends upon what you want to do with it. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
Geo [EMAIL PROTECTED] wrote: Anyone using inter Asterisk trunking IAX /IAX2 ? No - you're the first to think of that. Congratulations. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPComms Setup
Crystal Stream, Incorporated [EMAIL PROTECTED] wrote: Hey Moo. I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect attempt from 69.15.xxx.xx, request '[EMAIL PROTECTED]' does not exist Do you have a context called [voicepulse-in] containing a definition of what to do when someone calls that DDI number? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPComms Setup
Crystal Stream, Incorporated [EMAIL PROTECTED] wrote: this isn't working [IPComms-in] exten = s,1,Noop(${DATETIME} ${CALLERID}) exten = s,2,SetCallerID(${CALLERID}) exten = s,3,Answer exten = s,4,Goto(main-menu,s,2) exten = s,5,Hangup What I have is a block of 20 DIDs and I want to accept calls from all of them. It would be way to freaking complicated to do exten = 2027575120,1,Noop( . exten = 2027575121,1,Noop( et cetera How do I get this done? You could wildcard your DDIs, replacing 20, 21 etc. with [23][0-9], or whatever. Alternatively, you could create a macro that would look a lot like the body of your [IPComms-in] context, and then call that from 20 separate DDI exten lines. I'd just go with the wildcard. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Machine Detection
Anyone aware if Digium or Sangoma, or possibly a function of Asterisk, supports answering machine detection on an outbound call? I usually just listen to the voice. If it says something along the lines of leave a message after the tone then it's probably a machine. I can then choose to leave a message or just hang up. I imagine that most people would take a similar approach, even though the steps are probably not documented anywhere. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemailmain automatic extension detection?
Mason Loring Bliss [EMAIL PROTECTED] wrote: Is there a way I can have voice mail check calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for clues! Do you mean something like VoiceMailMain(${CALLERIDNUM})? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 patent in France
Amaury BOSSE [EMAIL PROTECTED] wrote: I am building an Asterisk PBX with voicemail and music on hold functions. An ISDN BRI line will also be available and G.729 IP-phones will be used. Are there patents rights applicable to France? The European Parliament recently voted 648 to 14 to reject the Computer Implemented Inventions Directive. The directive was supported by large monopolists such as Microsoft and would have thrown us into the same software patent minefield as the USA. The defeat of the bill means that individual EU member countries will continue to make their own decisions on what is patentable, rather than being hamstrung by the proposed EU-wide bill. Software-only patents are not valid in England and probably not in France, although that's for you to check. I understand that a limited number of software patents are valid in Italy. Software is protected by copyright, and that's enough. Ideas are free. http://www.nosoftwarepatents.com/ -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 patent in France
Steve Underwood [EMAIL PROTECTED] wrote: A large percentage of the patents applicable to G.729 are held by France Telecom. Now guess whether they bothered to get those patents in France. British Telecom has a large number of patents in North America. It can't use its software-only patents in England, of course. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: defunct email kill list
Dean Collins [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Is there an email address that we can forward 'defunct' emails like the one below to so that they can be taken off the mailing list. I appreciate it means work for someone but this email is probably being sent out to 400 'original' posters a day if that makes sense. It could also be the email address that we forward out of office emails to that could automatically suspend them for a period of time. You could probably automate the service but wouldnt be hard to have a person read the email, copy the address and paste it into a webpage. Just a thought. It seems to me that an automated solution to this problem would leave itself open to abuse. A moderated system would have to be monitored by someone, and that would seem to be far too much of a chore. You'll probably find that people with the appropriate access will unsubscribe or suspend accounts that send them OoO messages anyway, without further prompting from other subscribers. Perhaps your proposed auto-unsubscribe feature should be reserved for people who insist upon sending HTML emails and/or lazily top-posting their followups. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Looking for Web based SIP endpoint
Matt Riddell [EMAIL PROTECTED] wrote: Jason Becker wrote: https://sip-communicator.dev.java.net/ Don't know the current state of functionality with Asterisk. I couldn't get it to work many months ago - even with help from the developer. Any reason you are looking for SIP and not IAX? Is there an IAX alternative that you'd recommend? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Installing to a prefix.
Trevor G. Hammonds [EMAIL PROTECTED] wrote: Kevin Walsh wrote on Friday, 19 August 2005 6:58 PM: Andrew Kohlsmith [EMAIL PROTECTED] wrote: May I humbly request they be attached to a feature request on Mantis? I've been less than humbly requested not to do that sort of thing any longer, as I haven't signed a disclaimer. Sorry about that. The Asterisk change is trivial; Just set the INSTALL_PREFIX variable in the Makefile and then modify asterisk.conf and possibly musiconhold.conf. The Zaptel Makefile changes are a bit more involved. The diff file is 148 lines long. I've never had cause to look at libpri. How about submitting a disclaimer to Digium for the modified makefiles? Let's not bring that subject up again. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Installing to a prefix.
Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Aug 20, 2005 at 02:57:50AM +0100, Kevin Walsh wrote: Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 19 August 2005 21:27, Kevin Walsh wrote: I'll send the modified Makefiles to anyone who needs them. May I humbly request they be attached to a feature request on Mantis? I've been less than humbly requested not to do that sort of thing any longer, as I haven't signed a disclaimer. Sorry about that. The Asterisk change is trivial; Just set the INSTALL_PREFIX variable in the Makefile and then modify asterisk.conf and possibly musiconhold.conf. The Zaptel Makefile changes are a bit more involved. The diff file is 148 lines long. I've never had cause to look at libpri. INSTALL_PREFIX/DESTDIR is not instended for that. It is intended for installing asterisk to a different prefix than the one you build it to. It is commonly used for building installation packages (e.g: for rpms or debs). That's fine for Asterisk. The Zaptel Makefile, as distributed, doesn't play nicely if you change the prefix. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Installing to a prefix.
Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 19 August 2005 21:27, Kevin Walsh wrote: I'll send the modified Makefiles to anyone who needs them. May I humbly request they be attached to a feature request on Mantis? I've been less than humbly requested not to do that sort of thing any longer, as I haven't signed a disclaimer. Sorry about that. The Asterisk change is trivial; Just set the INSTALL_PREFIX variable in the Makefile and then modify asterisk.conf and possibly musiconhold.conf. The Zaptel Makefile changes are a bit more involved. The diff file is 148 lines long. I've never had cause to look at libpri. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: call load balancing
Anton Krall [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] intruder]# ps afx|more PID TTY STAT TIME COMMAND 1 ?S 0:08 init 2 ?SW 0:00 [keventd] 3 ?SW 0:00 [kapmd] 4 ?SWN0:00 [ksoftirqd_CPU0] 9 ?SW 0:00 [bdflush] No priorities.. Am I missing something? Try ps alx (Look at the NI column). Also see man ps. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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 2.0 Stable release out now
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Justin Selleck wrote: Is asterisk 2.0 real? Running in c#? I see references to it but cannot find it anywhere. r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. I think you might have replied to the wrong article. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: call load balancing
Joseph [EMAIL PROTECTED] wrote: On Wed, 2005-08-10 at 09:11 -0700, 1 2 wrote: I run asterisk with the -p option instead of messing with nice levels and it seems to make an improvement. If asterisk starts from the script how to append -p option. This is command from the script that starts asterisk: start-stop-daemon --start --exec /usr/sbin/asterisk \ ${OPTS} -- ${ASTERISK_OPTS} adding -p doesn't work: start-stop-daemon --start --exec /usr/sbin/asterisk -p \ ${OPTS} -- ${ASTERISK_OPTS} Put the -p in ${ASTERISK_OPTS} or at the end of that line. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] call load balancing
Joseph [EMAIL PROTECTED] wrote: On Wed, 2005-08-10 at 08:10 +0400, Jean-Michel Hiver wrote: I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable difference in call statistics (i.e. avg length of calls). If you are using ADSL, the maximum bandwith you'll be able to use is your upload rate since VoIP calls send data bidirectionally. Of course if you're using g.711 it's a different kettle of fish since it takes 80kbps (g.729 only uses about 24). According to Wiki: G729 is 8Kbps G711 is 64Kbps http://www.voip-info.org/tiki-index.php?page=Codecs That's the payload. You need to add the IP overhead to those numbers, which will bring the total to what Jean said, above. Trunking your calls over an IAX link will help reduce the total IP overhead. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: call load balancing
Joseph [EMAIL PROTECTED] wrote: Don't forget to experiment with nice to increase priority of for Asterisk. By default asterisk run with priority 0 same as apache and any other applications. We run a web-server on the same machine as asterisk and increasing nice for Asterisk to -15 helped a lot. You don't need to mess about with nice. Just run Asterisk with realtime priority; Use the -p switch when you start the Asterisk daemon. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Full T38 sip Faxing now Available
Carlos [EMAIL PROTECTED] lazily top-posted: Has anyone got a response from this? It was just spam. Forget it. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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.org beta site up!
Kristof Hardy [EMAIL PROTECTED] wrote: Matt Brooks wrote: I am just emailing to inform you guys that a new website has been created for asterisk.org. You can find the beta site up at http://beta.asterisk.org. It utilizes the drupal portal framework and Looking very good and much easier to navigate! Great work! Well, at least the new website doesn't say that I can register today to participate in an event that took place last June (see www.asterisk.org). -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Full T38 sip Faxing now Available
Chris Mason (Lists) [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Carlos [EMAIL PROTECTED] lazily top-posted: Has anyone got a response from this? It was just spam. Forget it. I have an account with them, just waiting for a suitable ATA to arrive. Good for you. Personally, I never buy anything from spammers. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Full T38 sip Faxing now Available
Michael D Schelin [EMAIL PROTECTED] wrote: Why do you put me down? I have not done a thing to you and I'm not a spammer. Please stop this activity It's not professional. If I were to give you bad service please feel free to comment negatively but I've never dealt with you nor do you have an account with us. As I understand it, you sent non-Asterisk-related commercial announcement to the Asterisk Users' mail list. What made you think that that wouldn't be considered to be Spam? I obviously can't comment on your service, as I'm unlikely to become a customer. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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-Dev] Digium to Sponsor a Pizza party atCluecon
Brian West [EMAIL PROTECTED] wrote: Digium, the creator and primary developer of Asterisk, the industrys first Open Source PBX, will be hosting a pizza party from 4pm to 6pm on the first day of Cluecon. We look forward to everyone coming out to enjoy this opportunity to meet fellow developers and users in a more casual environment. Thanks. I missed the first 600 copies of that announcement. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] help on linux version
amna saleem [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) This is amna saleem.I needed to ask if asterisk-1.0.3 can run on linux enterprise edition(latest version 4) I assume you mean Red Hat Enterprise. It should run on any modern GNU/Linux distribution. All of the servers under my control are running Gentoo. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Public phone
Chris Mason (Lists) [EMAIL PROTECTED] wrote: A client wants to put phones in a semi-public place, using a Calling Card solution. What kind of hardware is suitable? I'm looking for a wall mounted booth, a SIP phone that can't easily be broken, but I might just use cheap analog phones and a channel bank. What do you suggest for the calling card software? This installation will be outside the US. Buy a payphone with an internal rate table and hook it up to a Sipura SPA-2000, or a similar ATA box. That'll save a lot of messing about with calling cards etc. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Full T38 sip Faxing now Available
Tom [EMAIL PROTECTED] top-posted: Why the big secret? Why not post your solution to the list? It's probably just another one of those nasty closed source add-ons for sale. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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-Dev] Cluecon - Who's going ?
Greg Boehnlein [EMAIL PROTECTED] wrote: I'll be REALLY interested in your talk! Please make sure that you have take-away notes available so it doesn't evaporate into thin air after the conference! :) Perhaps also a drive-through lane at the side of the venue for people who don't have the time to find a seat, let alone sit and take notes. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Brian West [EMAIL PROTECTED] wrote: Or better yet.. modify the disclaimer like I and a few others did to say that the only thing you will disclaim are things you post on the bug tracker! NO UPDATES, NO CHANGES, NO NOTHING! If its not posted under your user on mantis IT IS NOT DISCLAIMED! That would seem to be a reasonable suggestion. That's what I did. I modified the disclaimer to only apply to stuff I post to bugs.digium.com under a specific userid. I did this to keep stuff I post to the mailing lists or on the web from being accidently disclaimed. Most people probably are not aware that that's an option. I certainly wasn't aware of it. If the owner accepts custom agreements, rather than just one of the two published versions, then that's a good start. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Brian West [EMAIL PROTECTED] wrote: Or better yet.. modify the disclaimer like I and a few others did to say that the only thing you will disclaim are things you post on the bug tracker! NO UPDATES, NO CHANGES, NO NOTHING! If its not posted under your user on mantis IT IS NOT DISCLAIMED! That would seem to be a reasonable suggestion. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Tzafrir Cohen [EMAIL PROTECTED] wrote: Disclaimers aside, who has the copyrights in those cases? Digium currently holds copyrights and/or is allowed to relicense the full asterisk codebase as is currently distributed in the asterisk tarballs on ftp.asterisk.org and also all the code in the asterisk CVS on cvs.digium.org (any better definition?) . Just to be clear, the perpetual agreement doesn't force a transfer of copyright; The author gets to keep the copyright, and can do whatever he likes with the code. The shorter disclaimer puts the copyright into the public domain. The perpetual agreement gives the owner two main rights. Firstly paragraph 1 allows the code, and all future Asterisk-related code, written by that contributor to be closed by the owner. Secondly, paragraphs 2 and 5(a) force the contributor to report all future changes and/or enhancements to save the owner the hassle of having to scour future forks looking for code that they might be interested in folding into their proprietary release. The second disclaimer (the short one) simply dumps all of your changes and enhancements into the public domain for anyone to use in a proprietary product. Of course, the only people who would know about this would be the signer and the company to which the document was sent. The short disclaimer is sufficiently woolly to allow for all future changes to a fork to be folded back into the binary release, although it doesn't include an obligation to report all such changes. Once signed, neither agreement has an exit clause or time limit, so neither of them can be cancelled. Maybe that is legal in Alabama (or Delaware), but I wouldn't really want to have to travel there to find out. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote: Adam Goryachev [EMAIL PROTECTED] wrote: On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote: For this reason, I believe that if a fork were ever necessary, it would struggle to beat a distinct path away from the Asterisk Binary Edition Correct, until the point where there is MORE features being added to the forked version of asterisk than the digium version of asterisk. That can't happen, because the ABE could, and probably would, absorb all of the advances in the fork, while forging ahead with the original. Since the fork would be GPL only, if ABE 'absorbed' the new features, then it would 'become' GPL, and therefore would need to be released as GPL, and hence would no longer by ABE :) So, that can't happen. Any other ideas? You're forgetting about the disclaimer documents. Anyone who signed the perpetual agreement and made changes and/or enhancements to the Asterisk code (a fork would still be using Asterisk code) would firstly be obliged to inform the owner, and would secondly have a prior agreement with the owner to allow them to use and close the code. That would neatly bypass the GPL and allow the new code to be folded into the Asterisk Binary Edition. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Kevin P. Fleming [EMAIL PROTECTED] wrote: Kevin Walsh wrote: The perpetual agreement grants the owner a non-cancellable right to use changes and/or enhancements made to the Asterisk codebase as [the] owner sees fit. As any Asterisk fork would, of course, be based upon existing Asterisk code, the owner would have the automatic right to take any code they wanted and backport it into the Asterisk Binary Edition - as long as the contributor to the fork had previously signed a perpetual disclaimer at some point in the past. Nice work clipping out only the words you wanted to use there! Let's try this again, with the actual text from the disclaimer: (b) The rights made in Para. 1(a) of this Agreement applies to all past and future contributions of Contributer that constitute changes and enhancements to the Program. 2. Contributer shall report to Owner all changes and/or enhancements to the Program which are covered by this Agreement, and (to the extent known to Contributer) any outstanding rights, or claims of rights, of any person, that might be adverse to the rights of Contributer or Owner. In other words, the _only_ code that the disclaimer covers is that which the Contributer directly identifies to Digium to be covered by the disclaimer. In absolutely no way does this disclaimer give Digium the right to appropriate other changes the Contributer makes to the covered programs without their knowledge and permission. Firstly, there are no in other words about it. That is a legal document. If other words are meant then they should be stated as such - in plain English. Secondly, paragraph 2 is distinct from paragraph 1, in which paragraph 2 insists that the contributor to a fork to also report changes back to the owner. If the owner doesn't report changes then they could find themselves in trouble over a breach of the agreement, and the owner can still simply take the changes anyway, as allowed for in paragraphs 1(a) and 1(b). In addition, even the most liberal interpretation of these clauses still includes the words Contributer and contribution, which clearly means that the entity signing the disclaimer has sole discretion which of their changes are covered and which are not. If that's the intention then it should be made clear in the document. The agreement, as it stands today, contradicts your statement and I believe it has been very carefully worded to either hide its true intentions or to allow future loopholes in favour of the owner. By the way, if anyone wants to see the full text of the dangerously perpetual disclaimer, they can find it here: http://www.digium.com/disclaimer.txt Read it very carefully, or have a lawyer advise you as to its content and implications. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Adam Goryachev [EMAIL PROTECTED] wrote: On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote: It has been flippantly said, a number of times, that if you don't like the situation then you can fork the project. A major fork seems (to me) to be pointless for one main reason (and a couple of lesser reasons): As I see it, anyone working on an Asterisk fork who had previously signed the dangerous disclaimer (the perpetual one) could find their changes to the fork rolled back into the Asterisk Binary Edition without any further permission being required. The perpetual agreement grants the owner a non-cancellable right to use changes and/or enhancements made to the Asterisk codebase as [the] owner sees fit. As any Asterisk fork would, of course, be based IANAL, but I assume you also have the right to revoke the agreement as relating to future patches. ie, it is non-cancellable in that I can't contribute something today, and next week change my mind. I am sure I can sign the agreement, contribute enhancements, cancel my agreement, and no longer contribute enhancements. That is not the case. The agreement makes it clear that 1(a) the signer does hereby grant, a non-exclusive, royalty-free and non-cancellable right to use changes and/or enhancements made to the programs. and 1(b) this Agreement applies to all past and future contributions of Contributer (sic). There is no provision to cancel, and furthermore, the signer specifically agrees to this arrangement by signing. For this reason, I believe that if a fork were ever necessary, it would struggle to beat a distinct path away from the Asterisk Binary Edition Correct, until the point where there is MORE features being added to the forked version of asterisk than the digium version of asterisk. That can't happen, because the ABE could, and probably would, absorb all of the advances in the fork, while forging ahead with the original. The *average* feeling of the community is that they are happy with the status quo. The status quo has been disrupted with the unveiling of the Asterisk Binary Edition. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Brian Capouch [EMAIL PROTECTED] wrote: Kevin Walsh and Aidan are able to see things that the rest of us cannot. Digium has duped you into associating with their evil enterprise to appropriate everyone else's hard work. I'm sure the stuff you and Mark have contributed pales in comparison with *their* contributions!! You'll never know. Contributors are required to sign a disclaimer, which is something I cannot do. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 22 July 2005 12:04, Lee Howard wrote: Well, I'm sure that was an added bonus. :-) Free work and free money. It reminds me of a certain Dire Straits lyric. Yes but are the chicks free? All except for the binary ones. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Steve Underwood [EMAIL PROTECTED] wrote: Kevin P. Fleming wrote: 2. Contributer shall report to Owner all changes and/or enhancements to the Program which are covered by this Agreement, and (to the extent known to Contributer) any outstanding rights, or claims of rights, of any person, that might be adverse to the rights of Contributer or Owner. In other words, the _only_ code that the disclaimer covers is that which the Contributer directly identifies to Digium to be covered by the disclaimer. In absolutely no way does this disclaimer give Digium the right to appropriate other changes the Contributer makes to the covered programs without their knowledge and permission. Well, yes, that's the idea. However, those of us who contribute don't generally provide any clear traceable definition of what we are contributing and what we are not. The documentation here is pretty woolly. As I pointed out, paragraph 2 doesn't have any bearing upon what code the owner is allowed to incorporate into the Asterisk Binary Edition; That non-cancellable right is granted in paragraph 1. The quoted paragraph (2) simply forces the person who signed the document to report any changes to the Asterisk codebase. In theory, even minor changes that would be of no use to the wider community must be reported to the owner, who would make the final decision. A fork followed by changes and/or enhancements to the code would be covered by that paragraph - forever. It's all very underhanded. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Last two digits getting cut off?
Rob Engstrom [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) We've just setup our [EMAIL PROTECTED] server, with our quad port card. Everything works well so far. One thing I notice is that when I leave the handset on the hook and dial a #, all is well. If I pick up the phone and dial, it cuts off at 10 digits, which is a problem if I need to dial 1+area+phone # (12 digits). The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a config to allow more than 10 digits? The Cisco 7960 and Sipura devices have a dialplan that define when to send the dialled number to the server. If the SoundPoint has something similar, and I expect that it does, then that would be a good place to start looking. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Business Edition
Lee Howard [EMAIL PROTECTED] wrote: Kevin P. Fleming wrote: You seem to be neglecting the amount of work that Digium puts into the Asterisk (and related) products on an ongoing basis that is given to the community at no charge. So at least we agree, then, on what the reasoning is. Digium feels that the community owes it to them. I agree with that assessment. It has been flippantly said, a number of times, that if you don't like the situation then you can fork the project. A major fork seems (to me) to be pointless for one main reason (and a couple of lesser reasons): As I see it, anyone working on an Asterisk fork who had previously signed the dangerous disclaimer (the perpetual one) could find their changes to the fork rolled back into the Asterisk Binary Edition without any further permission being required. The perpetual agreement grants the owner a non-cancellable right to use changes and/or enhancements made to the Asterisk codebase as [the] owner sees fit. As any Asterisk fork would, of course, be based upon existing Asterisk code, the owner would have the automatic right to take any code they wanted and backport it into the Asterisk Binary Edition - as long as the contributor to the fork had previously signed a perpetual disclaimer at some point in the past. A fork wouldn't get very far without the support of at least some of the regular contributors, all of whom have probably signed the perpetual agreement. For this reason, I believe that if a fork were ever necessary, it would struggle to beat a distinct path away from the Asterisk Binary Edition, which would be free to assimilate any advances into its own codebase. To mitigate the above, I believe that the perpetual disclaimer should be modified to cover only a specific time period (i.e. one year from the date of submission). All contributions made within that time period would be covered by the currently-valid agreement, and that agreement could be renewed annually, if desired. I don't see that sort of change happening anytime soon because I believe that the perpetual nature of the agreement is quite deliberate. I think people sign agreements out of convenience, or pressure, without reading carefully enough. For instance, I wonder how many people actually received their $1.00 (One Dollar) and other good and valuable consideration when they signed their future options away. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Business Edition
Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Monday 18 July 2005 23:50, Kevin Walsh wrote: I suspect that there is now less of an incentive to produce stable branches, and backport fixes to those branches from the development version, as this could possibly reduce the value of the closed version somewhat. It could turn out that we eventually find the project in a permanent in development state, with no stable releases at all - just the CVS HEAD. Once you start down that road, and rely upon revenue generated from closed source products, it's difficult to turn back. This is a typical slippery slope argument. There has been no indication that this is what is occuring, nor that this is what will occur. I personally suspect that at some point the 1.0.x series of Asterisk and ABE will end up being the same thing; a feature-frozen version of Asterisk. ABE might have some kind of checksum or other authentication/verification wrapper to ensure that it's actually ABE and not almost ABE. Well, we can only hope so. I suspect the opposite and, so far, all of my negative fears and suspicions have come to pass. It was very easy to foresee that the disclaimer documents' sole purpose was to allow for the future closing of the source. This has been pointed out and warned against for years, and now it is sad to see it starting to happen. These are very worrying times for the future of the project. If you don't want or don't like ABE, don't use it. Nobody is cramming it down your throat. That's not the point. What exactly is the point, then? You don't like it because it's not free as in libre, but you don't propose any solid way for Digium to maintain its profitibility and grow in order to help support the community and make Asterisk grow. I understand that you want Asterisk totally free but I don't see a way to do so, and I don't think you do, either. Digium can remain profitable by selling hardware and support. I suspect that if someone using the GPL version was to contact Digium and offer to pay for support, they wouldn't be turned away for not using the Asterisk Binary Edition. The ABE allows limited support (whatever that means) to be paid for in advance, possibly in the hope that a lot of people won't actually use all of their allowance. That's a good strategy, but one that would still be possible with the GPL version. I define free as libre, as you said. This doesn't rule out the possibility of charging for installation, support and customisation, nor making money in other ways. The Asterisk Binary Edition isn't free in any sense of the word. You've presented a slippery slope argument and a strawman (disclaimers) and I am interested in finding out how you'd do it if it were your ship to steer. You're not a screaming zealot and I think this discussion's good for the list. If it were my project to steer, then there would be no disclaimers, and therefore no possibility of a non-GPL release. I believe the project would move forward a lot quicker if everyone was given the opportunity to offer code and patches under the terms of the GPL, rather than simply relying upon a few committed developers who are happy to donate their code to a single company to close and sell. Don't get me wrong. I use and recommend Asterisk, and I think the software is great - a credit to the dedicated individuals who keep it going. I just don't like the one individual company which constantly tries to claim all of the credit for what is clearly a community effort. One piece of good news can be found here: http://www.asterisk.org/index.php?menu=summer_of_code The requirements say nothing about being asked to sign a disclaimer, so perhaps either Google have views on this sort of practice, or people will be quietly rejected, during the interview process, based upon their willingness to have their source code closed. I suspect the latter, which would not make it good news after all. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Installation
Aron Bereket [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) I am a new user of Asterisk. When I downloaded it and was trying to compile it, i got the following error message after it run for a while. This error came after I run make. bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c ast_expr.y:110: unrecognized: %locations ast_expr.y:110:Skipping to next % ast_expr.y:141: invalid @-construct ast_expr.y:141: $. is invalid ast_expr.y:141: invalid @-construct I suspect that you're running an ancient version of Bison. If that's the case then upgrade it and try again. Start your day with Yahoo! - make it your home page Why? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Business Edition
KRTorio [EMAIL PROTECTED] wrote: For $995+ (including support), a technical manual, and scripts, is it worth switching to the business edition? Absolutely not. If you find that you need $995 worth of support, some time in the future, then I'm sure that you can obtain it from one of several providers. I don't think it's worth paying up-front for something you probably won't need, but that's really for you to decide. If you really want to pay $995 for a closed source product with some features removed and license control added, then go for it. It's your money. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Business Edition
Andrew Kohlsmith [EMAIL PROTECTED] wrote: I dunno... people seem all up in arms about this but honestly I fail to see the problem. Digium is doing what they can to make money and provide services while keeping Asterisk as free and openly developed as possible. Services could be provided, and money could be made, without resorting to selling closed source versions of the product. Apparently, the closed version consists of the contents of CVS HEAD, with various changes made to increase reliability and decrease risk - according to the FAQ. It would be nice if the binary version's source was available as a branch in CVS, but that probably doesn't fit into a closed source business model very comfortably. I suspect that there is now less of an incentive to produce stable branches, and backport fixes to those branches from the development version, as this could possibly reduce the value of the closed version somewhat. It could turn out that we eventually find the project in a permanent in development state, with no stable releases at all - just the CVS HEAD. Once you start down that road, and rely upon revenue generated from closed source products, it's difficult to turn back. I have (small amounts of) code contributed to Asterisk and I am working on more. Digium and Asterisk have given me a lot of newfound freedom and flexibility and power in my phone system. I appreciate that, and I don't feel that this dual-licensing or granting of a nonexclusive perpetual license to the bits and pieces of my code is too much to ask. My bits and pieces would be worthless without the bits and pieces and chunks and slabs of code that others have provided, and it'd all be useless without the framework that Digium came out with. Asterisk would not be the product it is without the efforts of the community who, it seems, have provided the majority of the source code and support for the project. Of course, Digium try their best to not accept patches to their code unless they are accompanied with a disclaimer. According to the bug tracker (http://bugs.digium.com/main_page.php), the disclaimers are insisted upon in order to keep copyright clean, even though it has been pointed out, several times, that the agreements have no effect on copyright at all. The disclaimers exist to grant Digium the right to close and sell your code. If you're happy with that then that's your choice to make. If you don't want or don't like ABE, don't use it. Nobody is cramming it down your throat. That's not the point. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] G.729 licensing - HardwareDevices rather than software
trixter [EMAIL PROTECTED] wrote: On Mon, 2005-07-18 at 21:45 -0600, Tim Pushor wrote: Just gotta watch that you dont have two with the same mac addr in some networks (some systems and network devices dont care enough others completly come unglued). Yeah, like ethernet. let me clarify, on an ethernet network some systems and devices dont care others freak out. A better example would be the case where two machines are on different Ethernet networks. Perhaps two PBXs, in two separate offices, connected via the Internet. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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 timing device
Umar Sear [EMAIL PROTECTED] wrote: I have been using asterisk for some time now. However I have never used it with any of the digium or compatable cards (Purely used for SIP). I understand that for using Meetme, I need to have a timing device, which could either be hardware or zrdummy etc (I am not using any right now). There's also zaprtc to consider. See here: http://www.voip-info.org/wiki-Asterisk+timer Can someone tell me if the timing device is needed for voicemail and other applications too?. No - the timer is only required for conferences, IAX trunking and MOH. There might be others I've missed, but Voicemail is not one of them. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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 UK Community
Ben Merrills [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) To those who are/were interested J Ok, Ive setup a mailman mailing list. I dont so this often, so Ive done a bad job of it, please let me know. Oh, and I know the SSL cert has expired was self signed J Now Ive apologised for my crapness, the URL to signup to the mailing list is: If the list founder posts using HTML then that doesn't bode well. I think I'll give it a miss and continue to use the main mail lists. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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-Dev] RE: [Asterisk-biz]Asterisk training andcertification :: AstriconTraining
...For Asterisk gurus, that believe that you can take the exam without attending the training, there will be exam oppurtunities setup in combination with Astricon conferences. When we update dCAP for future releases of Asterisk (1.1, 2.0), you will be able to upgrade your certification at Astricon... Great - so that's another $3000 every few months then. I'll just laugh at anyone who's stupid enough to show me their certificate. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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 server to asterisk server
William Betts [EMAIL PROTECTED] wrote: what is the best way to have 2 asterisk servers communicate with each other? Probably using IAX2. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] soho usage
Mathias Houngbo [EMAIL PROTECTED] wrote: i want to know if it is possible to use an analogic phone with a modem in asterisk ? i don't want to use a digium card or any FOX/FXS module ! perhaps a modem card ?? You could consider a Sipura device. The SPA-2000 comes with two FXS ports, for two analogue phones, and the SPA-3000 comes with one FXS and one FXO port. Phone ports on modems usually just pass through to the phone line, so I doubt that you could use anything like that to set up an analogue phone as an Asterisk extension. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] SOHO PBX using asterisk
Christopher L. Wade [EMAIL PROTECTED] wrote: Oops, just noticed, for 5 (FIVE) phones plus using the analog line related to your ADSL line, you'd need a TDM40B and a TDM11B. Or two Sipura SPA-2000s plus one SPA-3000. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] SOHO PBX using asterisk
Steve Wolfe [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Or two Sipura SPA-2000s plus one SPA-3000. Can you tell us more about the sipura device? A Sipura SPA-2000 provides two FXS ports for two analogue phones. You can use a bunch of these devices if you need more ports. Of course, once you get to a certain number, a channel bank will be cheaper. A Sipura SPA-3000 provides one FXS port, as above, and one FXO port for your phone line. The configuration quoted above (2 x SPA-2000 and 1 x SPA-3000) will provide 5 x FXS ports and 1 x FXO port, and should cost less than $350 USD. Also if one is using sipura devices I would still be tempted to put in a real digium card -- or how well the the ztdummy work for clocking? If you need a hardware timing device for some reason then a cheap X100P will do the trick. The software timer modules should be ok, on a Linux 2.6 kernel, for most uses. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] modified prepaid application
Adnan Ahmed [EMAIL PROTECTED] wrote: how can we integrate the modified-prepaid application with asterisk because when i compile the app_prepaid it gives bunch of errors. I have no idea. Perhaps you're out of disk space. No? Ah well - my mind reading powers seem to get weaker over long distances. Posting the error messages might help. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] ATA Adaptor
Jason Lane [EMAIL PROTECTED] wrote: I am new to asterisk and I am trying to get things set up so I can prove to the boss it works and get the budget to do a full implementation. Does anyone have an ata adaptor or an ip phone laying around they would be willing to sell me for around 30-50 dollars, I will need 2 of them. A Sipura SPA-2000 will cost you around $100 USD, brand new, and will provide you with two FXS ports. That should be enough for a quick demonstration. I don't know whether you were expecting each ATA to provide one FXS port or two for your $30-50 USD budget. If you don't get any offers from the subscribers of this mail list, you could try eBay for a second-hand ATA device or two. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] unsubscribe
Bart de Wild [EMAIL PROTECTED] wrote: unsubscribe No. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOC NearYou!)
Christopher Dobbs [EMAIL PROTECTED] wrote: My company has started development on a Ethernet based channel bank. Here are the (current) spec's - 10/100 Ethernet Port - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired) - Serial Console - TDMoE - IAX2 - EETP (A protocol that we have designed for IP Telephony) How does your proprietary EETP protocol differ from the proprietary IAX2? I assume your protocol has support for trunking which, it seems, is one of the main reasons why people use IAX2 instead of the SIP standard. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list broken again?
Greg - Cirelle Enterprises [EMAIL PROTECTED] wrote: It's been hours since I've seen a post from this list Must be broken again. You determined that the mail list was broken and decided to alert everyone by posting an article to the list? Good thinking. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conjuring Kevin Walsh (was: four wildcards ina single pc)
Gregory Junker [EMAIL PROTECTED] wrote: This is a good example for the newbies of the list as to why proper formatting and list ettiquitte is important. I made a mistake, it was easy enough for someone to come around behind me and correct the message. We all can make mistakes. Oh whatever, get off it already. Any minimally intelligent amoeba Would have understood the correction regardless of where it occurred in his post. This case is hardly the poster child for bottom-posting vs top-posting. Do you really want to start this nonsense up again? He probably doesn't, but it seems that you do. Why is that? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk from CVS
Adi Linden [EMAIL PROTECTED] wrote: I admit that this might be some very basic question... How do I obtain Asterisk 1.0.3 from CVS? Does '-r v1-0' get me 1.0 or 1.0.3? That will get you the very latest version of the 1.0 series, whatecver that might be. The version I have here shows me this: Asterisk CVS-v1-0/2004-12-10/18:45:43/cursor-5, Copyright (C) 1999-2004 Digium and others. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Door buzzer.
Cian O'Sullivan [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. If you need an FXO port and don't want to install a whole new server then you could consider an external device, such as a Sipura SPA-3000. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Hardware
Walid Azab [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Can I start using Asterisk with a couple of SIP IP phones and Softphone software on users PCs only? I do not have any cards yet and will still have to wait until I order a card. Yes. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Users list.
David Uzzell [EMAIL PROTECTED] wrote: Does this sudden rush of email mean we are all back online? No. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 algorithm?
Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: according to what I've found out this far, the G.729 patent seems not valid in a broad range of countries. That is correct. does anyone know where I can find the algorithm? G.729 consists of a lot of patents and a lot of algorithms. A couple of Google searches will scare up most of them. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 algorithm?
Robert Rozman [EMAIL PROTECTED] lazily top-posted: do you have info in what countries g.729 is not valid... ? You could start with the whole of Europe and can also add the UK. I'm sure there are lots of other countries who don't feel the need to acknowledge US-based software and algorithm patents too. This subject has been covered several zillion times in the mail list. Google is your friend. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Gentoo and Asterisk - any experiences?
Niels Chr. Sørensen [EMAIL PROTECTED] wrote: In constant search for optimization, a friend told us about his experience with Gentoo Linux-distro. He claimed that he doubled the performance of his server by changing to Gentoo from Debian. Does anyone have any experience with running Asterisk on a Gentoo linux? I use Gentoo - both at home and at work. You would see a vast difference between an old 2.4-based distro and Gentoo running 2.6, but a lot of that would be down to the scheduler in the 2.6 kernel. If you compile from source then there's no doubt that you'll get a speed boost. I doubt that you'd double the performance without other factors deserving most of the credit. I do recommend Gentoo GNU/Linux with the 2.6 kernel. Everything just works (tm) without all the hoops you have to jump though with other distros - such as Fedora. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Gentoo and Asterisk - any experiences?
Patrick [EMAIL PROTECTED] wrote: On Tue, 2004-11-30 at 16:23 +, Kevin Walsh wrote: I do recommend Gentoo GNU/Linux with the 2.6 kernel. Everything just works (tm) without all the hoops you have to jump though with other distros - such as Fedora. Not wanting to start the ol' distro wars again but I haven't come across any hoops I need to jump through while installing Fedora Core 2 or 3. My installs typically take 20 mins (start-end, including extensive individual package selection) which imho beats the lightyears it takes Gentoo to compile all the apps. Fedora Core 2 has worked fine for me running Asterisk for a long time now and I recommend that over Fedora Core 3 due to the introduction of udev which requires additional tweaks and too cutting edge versions of gcc for my taste. Additional tweaks are classed as hoops to jump through in my view. You're right about one thing though: We don't need another distro war. By the way, a light year is a unit of distance - not time. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'
Leandro Morgado [EMAIL PROTECTED] wrote: On Wed, 2004-11-24 at 13:53, Dave Cotton wrote: On Wed, 2004-11-24 at 13:45 +, Leandro Morgado wrote: Jose Hernandez wrote: I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. This is not udev up to its tricks again? You are right! If his distro is using udev it might dynamically create the zap devices with the wrong set of permissions. I am not using udev (debian 2.4) but it makes sense that udev would allow you to specify the permissions to use when creating devices! The following seems to work (udev permissions file): # zaptel device zap/*:asterisk:asterisk:644 zap/channel:asterisk:asterisk:660 zap/ctl:asterisk:asterisk:660 zap/pseudo:asterisk:asterisk:660 zap/timer:asterisk:asterisk:660 If you run Asterisk as root then the above is irrelevant, but nobody does that - right? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using IPKall and SIP with insecure=very
Rob Emanuele [EMAIL PROTECTED] wrote: I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip debugging turned on I see the call come in and the message below is printed. If I put the exten route that I have in the ipkall-inbound section of extensions.conf (below) into the default section it works fine, but isn't neat and elegant. How do I make incoming call from ipkall match a sip.conf section? From sip.conf: [3501] type=peer host=dynamic dtmfmode=rfc2833 context=ipkall-inbound insecure=very nat=no From extensions.conf: [ipkall-inbound] exten = 3501,1,Goto(menu,s,1) You'll probably find that there's no need to set up a specific user for IPKall. You were using type = peer, which would have been wrong anyway. In your [general] section, create a context = incoming-sip (or whatever you want to call it) and then set up a matching context in extensions.conf. Your extensions.conf context can then match your 3501 extension, along with any other direct incoming SIP addresses you need. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using IPKall and SIP with insecure=very
Rob Emanuele [EMAIL PROTECTED] wrote: Kevin Walsh wrote: You'll probably find that there's no need to set up a specific user for IPKall. You were using type = peer, which would have been wrong anyway. In your [general] section, create a context = incoming-sip (or whatever you want to call it) and then set up a matching context in extensions.conf. Your extensions.conf context can then match your 3501 extension, along with any other direct incoming SIP addresses you need. What if I wanted to create different incoming-sip contexts depending on the service being used or number being called? For example sip calls coming from ipkall goto one context that presents a menu, but another sip call coming from one of the free German services provides a different menu and in German. Sip.conf is quite flexible and can be set up in lots of different ways. If you had multiple accounts with multiple providers then you'd want to set up separate user/peer sections for each provider, and could easily set up each section to make use of its own specific context. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK available SIP phone?
Bill Seddon [EMAIL PROTECTED] wrote: I use Asterisk at home and have bought a couple of HandyTones ATAs. The DECT phones are plugged in and work really well. The ATAs are £56 from Goods2World (though the additional one I've just bought didn't work and is being returned) and about the same from VoIPTalk (who are out of stock currently) The only downside to the ATA is that I've not yet worked out how to have CallerID displayed on the DECT phones. A Sipura SPA-2000 would be cheaper than two HandyTone ATAs, and Sipura support the Caller ID name/number display on DECT phones without any problems. You can get one from CallUK for £85+VAT (http://www.calluk.com/sipura/). -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] i swtiched to digest
FuturaHost.Com Lists [EMAIL PROTECTED] wrote: I believe the list is so big that many of us are loosing some interesting threads. May be the admins can split the users list in some more specific sub-lists, and the people who wants to receive all the messages can subscribe to the sublists, or have a digest for someones, etc. You'll find that many people will want to be subscribed to all of the mail lists - just in case something interesting is said or asked. You'll also find that some Muppets will post their questions to multiple lists, instead of finding a specific list to use, or will judge that their question/comment has relevance in multiple lists. For instance, how many Asterisk veterans are likely to hang out on asterisk-newbies so that they can answer the same FAQ question every ten minutes, and how many Asterisk users are going to post their questions to the newbie list when they find that there are no experts there? People already post and/or duplicate end-user questions to the developers' list as it is. Splitting up the mail lists will most likely result in more bandwidth use for most people - not less. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] rtp codec error
Daniel Eboa [EMAIL PROTECTED] wrote: Hello all, And that's as far as I read. You should try re-posting your question without the HTML, without the bold multi-coloured text and without the pointless, out-of-focus graphics. Having done that, perhaps someone will be inclined to read your article and might even be able to help. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] i swtiched to digest
Chris TenHarmsel [EMAIL PROTECTED] wrote: I've ammased 12Mb of email in a weekthis list does generate a LOT of traffic that it gets the point for me that it's really hard to wade through it all. Just do what I do: 1. Scan the subject list and delete anything that doesn't appear to be of interest. 2. Delete any messages that have lazy, top-posted followups. A rule that marks articles that include -Original Message- in the text will find a whole load of these, and will make life a lot easier. Others will be found and deleted as and when they are opened, assuming they passed rule #1, of course. 3. Delete all HTML articles. I don't have a rule for these at the moment, so I usually just delete them as and when I find them. Following the above, you'll find that you're left with only 10% or so of your original mailbox size. This 10% usually results in the articles that are worth reading. Of course, everyone will have their own way of trimming public mail lists to a manageable size. This is mine. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Re: i swtiched to digest
FuturaHost.Com Lists [EMAIL PROTECTED] wrote: Anyone wanting to split the list in more parts? Yes and no would suffice, so we can close this without a talk long a year, and without someones forcing their point of view to others. No. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded
Gregory Junker [EMAIL PROTECTED] wrote: One problem is that the SPA3K only uses two-stage dialing on the FXO -- VoIP2 path, which means any time someone calls the phone system and gets forwarded to a select SPA3K extension, they hear a dial tone. As far as I can tell, there is no way to disable that. You can have it execute a particular dialplan in the SPA3K but the caller gets to hear the digits as they are dialed into Asterisk. You didn't say which firmware version you have, but I suspect that it's out of date. A quick update should fix that. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 for *
Matthew Crocker [EMAIL PROTECTED] wrote: Here's a question: if the author has purchased a commercial license to use Asterisk, and I get binary modules from him, I can still use them with my CVS-based Asterisk, right? You may be able to do that. You could always run a couple Asterisk boxes, run IAX2 between them and leave the commercial stuff with SS7 in stock (aka supported) configuration. I just avoid people who think it's ok to create proprietary extensions to free software. People like that should be ashamed of themselves, as it's just an insult to the people who have freely contributed to the project. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Top posting
Matt Riddell [EMAIL PROTECTED] wrote: Stephen R. Besch wrote: This all reminds me so much of Jonathan Swifts bit about the BigEndians and the LittleEndians (referring to which is the 'correct' end to open a soft boiled egg) in Gulliver's travels. But that's simple, surely you should put the big end of the egg into the egg cup and open from the tapered end, so as to avoid the egg falling over and losing its contents! :-) Of course, some people wouldn't spot that one end had already been opened and would proceed to open the other end as well. Unfortunately, these idiots usually end up making a complete pig's breakfast of the whole thing while irritating the more civilised diners. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The Apperiant Death of IAXtel
Christopher Dobbs [EMAIL PROTECTED] wrote: Given that IAXtel has not been responding for some time, I am willing to setup accounts for thoes who want to have that kind of functionallity. If you are interested, send me an email with your requested username and password, and i will send you your account information. Given that IAXtel has not been responding for some time, what use is a username and password? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Top posting
Paul Fielding [EMAIL PROTECTED] lazily top-posted: Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting. As you seem to find it difficult to move the cursor on your own, perhaps this utility will help: http://home.in.tum.de/~jain/software/oe-quotefix/ You could install it to fix your broken mail reader - if it's not too much effort. When you bottom post, I need to scroll way down the message to see your response The effort involved is clearly too much for you to handle. Are you really that lazy? If I want to see the source message *then* I'll scroll down, but chances are I've already been reading the thread so this isn't necessary. Your laziness will make life difficult for people who find your followups in a future Google search. Just because you've read the entire thread, doesn't mean that someone else will have done the same next year. Then again, the chance of you posting useful information for someone to find in Google does seem to be a bit remote. just my 2 cents That might be all your time is worth. Others get paid a little more than that. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: getting callerid from spa3k to asterisk
Randy Bush [EMAIL PROTECTED] wrote: if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. i found the problem, or at least a work-around. if i reverse the order of the above two sip contexts, the incoming call is properly routed to the spa3k-in sip context as opposed to the wrong one, spa3k-out. my guess is that * is traversing a list and taking the first context which has the ip address and port it wants without checking the context name against the name which was received over the wire. so it depends on what order the contexts are inserted in the list. Yes, but it's not a bug. :-) You have type=friend and type=peer. Change friend to user for the incoming definition. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Top posting
Bill Seddon [EMAIL PROTECTED] lazily top-posted: No, he didn't. But I'd guess that readers of his email who are, perhaps, less sophisticated users of email lists will have been concerned about the attitude he projected and be unnecessarily anxious about posting questions in the wrong way. Since there is no wrong way I just wanted to try to project a more flexible attitude. Perhaps you should adopt a more flexible attitude and learn to follow up properly. There is no excuse at all for lazily top-posting. There is a wrong way, and your method of posting demonstrates it clearly. Please locate your mouse, arrow keys or whatever you use to move your cursor around, and make an effort to post your followups in context. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Motherboard whitelist (was Echo - UK Impedanceproblem with X100P?)
On November 12, 2004 09:46 am, Rich Adamson wrote: As far as the motherboard issue, its not just the digium products that have an issue. If you know of someone that is heavy into using audio applications (eg, song writers, midi stuff), they have known about the pci / interrupt latency issues on certain motherboards for a lng time. Wouldn't doubt those folks maintain a list of what's reasonable verses unacceptable. You bring up an excellent point -- Can we use the blacklists and whitelists the audio folk have to help the asterisk community? They'll have all hte same issues -- shared interrupts, craptastic PCI chipsets, etc... sounds like a great idea to me. Why not just get a Sipura SPA-3000 and side-step the motherboard vs. TDM card compatibility issue altogether? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Top posting
Richard Cook [EMAIL PROTECTED] lazily top-posted: Must be nice to have time and money to worry about someone's posting method. Amazing. Time is money, and I will waste neither trying to work out what it is you just followed up to. Perhaps if you learned to post in context... -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK BT Caller ID, X100P and Asterisk v1
StrUK [EMAIL PROTECTED] wrote: Having one card with FXO and FXS modules on and consigning the X100P to the heap is a real goal - I need IRQs! Get a Sipura SPA-3000 and forget all about IRQs. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P CLONES again
Seth Remington [EMAIL PROTECTED] wrote: On Tue, 2004-11-09 at 15:22, [EMAIL PROTECTED] wrote: What model of modems can be use as X100P? I can get a Motorola 62802-52, did anybody ever try it? I must buy some of the clones because in my country nobody sells anything of voip yet. Check the Generic heading: http://www.asterisk.org/index.php?menu=hardware To see the product prices, call our Sales Department now at 999 and ask for the password. - Ha! If they're too embarrassed to display their prices then just avoid them. Why should you have to put in extra work just because they couldn't be bothered to put a proper e-commerce system in place. You'll find lots of alternative X100P products if you look around in Google and eBay. They are all clones of the same design - even the Digium product. Some are just more expensive than others. You should also check whether the Sipura SPA-3000 is technically compatible with the PSTN system used in your country. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Generic Boards - Low Prices / High Quality.
Richard Moore [EMAIL PROTECTED] wrote: As announced before we're starting selling only E100P based boards. T100P boards and other digium-like products will be available in 20-30 days. All products are being tested for our engineers. We're developing our website with all products and we'll include a web shopping for global sales. At this momment , i'm at Brazil negotiating a distribution channel for latin america. 500 units were sold only today. (E100P) So, please wait for our products to be released.! don't waste money :) Although your products are probably of interest to a large number of Asterisk users, you'll probably want to move your adverts onto the asterisk-biz mail list before people start whining. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E100P - Generic (Clone) - :)
Tom [EMAIL PROTECTED] wrote: So Richard, since you are a Asterisk Engineer, will you take over Asterisk software development if Mark Spencer cannot sell hardware to stay interested? You probably failed to notice the Asterisk community. Asterisk is larger than one man or company. Some Asterisk developers even post to the mail lists and answer questions etc. Competition is a good thing. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P CLONES again
Richard Lyman [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Seth Remington [EMAIL PROTECTED] wrote: Check the Generic heading: http://www.asterisk.org/index.php?menu=hardware To see the product prices, call our Sales Department now at 999 and ask for the password. - Ha! If they're too embarrassed to display their prices then just avoid them. Why should you have to put in extra work just because they couldn't be bothered to put a proper e-commerce system in place. You'll find lots of alternative X100P products if you look around in Google and eBay. They are all clones of the same design - even the Digium product. Some are just more expensive than others. You should also check whether the Sipura SPA-3000 is technically compatible with the PSTN system used in your country. maybe it's you who should be embarrassed. you see, they are a distributor, and therefore you don't get an account without a resale license. As all of the other links on that asterisk.org page are for retail sales, perhaps the so-called generic link should point at a retail website. As I said, a quick look on Google will reveal plenty of them. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK BT Caller ID, X100P and Asterisk v1
Richard Hamnett [EMAIL PROTECTED] wrote: Get a Sipura SPA-3000 and forget all about IRQs. How about just enable kernel APIC support and forget all about IRQs!! I thought everyone was using APIC now anyway. Why would someone not have it enabled? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: CallerID for the UK
Charles Osstyn [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Hi all, I am too new with Linux, to really experiment with the callerid. I know the problem is due to BT using a different technical platform. So as too using Perl or any other scripts and a modem to create the work around required to get the callerid to work UK I need some help in this field. Anyone, got a step-by-step guide, how to add this to a working setup? If you're using an X100P then I have some patches that will give you UK (BT) Caller*ID without having to resort to the use of modems and scripts. Email me off-list if you want the patch files. This would solve my last technical issue. Many thanks in advance also any one got any RPM's for a GUI which can be used to setup SIP and IAX account and configure the dial plan through a nice web interface. There are several web-based interfaces out there. Whether you consider any of them to be nice is up to you. :-) [snip: 40 line signature, unnecessary attachments and adverts] -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linux and Windows
Jay Milk [EMAIL PROTECTED] lazily top-posted: I'm not sure why I'm even discussing the benefits of one operating system over another I don't know why you bother posting at all. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users