[asterisk-users] taskprocessor.c: The 'sorcery/contact-00000015' task processor queue reached 1500 scheduled tasks.
Running 15.1.2. I have four devices transitioned to use pjsip. After about 1-2 days of uptime, psjip stops accepting registrations, and the messages log contains the entry as per the subject. At any given time, "pjsip show contacts" only shows the four devices. Could someone point me to a fix, short of rebooting the server every day? Thanks. b. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP console messages with Zoiper
I'm running Asterisk 15.1.0 and in the process of converting my various SIP endpoints to use PJSIP. My Zoiper client causes the messages quoted below to show up on the CLI once per minute. Things seem to work OK, but I am curious because there seems to be no way to suppress the messages, and there are three per minute, clogging up the console. Thanks. b. ** snip ** -- Added contact 'sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096' to AOR '8005' with expiration of 60 seconds == Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096 has been deleted == Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096 has been created -- Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096 is now Unknown. RTT: 0.000 msec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior?
The subject says it all. I'm betting there's a way to do it, but so far I haven't found the dialplan runestone via web searching. Thanks. b. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Exceptionally long queue length queuing . . . .
I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. This problem began for me over a year ago, and continues up to the latest versions I've installed (1.6.2.13). It happens randomly, and the suggestion on one of the bug tracker tickets that it is instigated by a small network leg looks to be on point to me, because while it happens way often, it doesn't always happen. My ITSPs have all dropped IAX, and if they're experiencing this problem I can see why. Once the first of these messages has occurred, it's goodbye audio for the rest of the call. If anyone has a perspective on this longstanding problem, I'd sure be glad to hear it. Thanks. b. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Alex Balashov wrote: ) How about rejecting emails that don't have a subject? That is an excellent idea. If a person doesn't have enough clue to use a subject, then we're really just feeding the beast when we indulge the question with an answer. And the archived version of that question/answer are pretty useless, too. Thx. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need * consultant in houston area
Could you all please take this COMMERCIAL discussion to the -biz list? Thanks. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gemeinschaft released
Philipp Kempgen wrote: The cluster capability is very interesting. Exactly. Although I must admit wo don't really have much documentation on how to install a Gemeinschaft cluster. Is there anything documented in English? Philipp's mail doesn't say what the product does, and the website is in German, which I don't read. So I'm wondering if I missed a discussion of what it does in the first place! Thx. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk content @ OSCON 2008?
Martin Smith wrote: Hey folks, Is anyone working on Asterisk (or other) presentation proposals for OSCON 2008 in Portland, OR? Here's the link, in case: http://en.oreilly.com/oscon2008/public/cfp/13 I'm working on a proposal to do a tutorial there on Asterisk. I have done such the past three years. I'm also proposing a session on embedded Asterisk, under openWRT, which is my little fringe niche of the world. . . B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime: Should I say or should I go (now) ?
Terry Wilson wrote: And a free hint: if you are going to have to do anything that resembles number porting, swapping extensions, etc.--don't use extensions/phone numbers as SIP usernames. You have to regenerate config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. I second that emotion. I consult with a bunch of people who rolled their own Asterisk systems long ago, and when the try to virtualize their system in various ways they find their hands are tied. It is indescribably confusing once the number in sip.conf gets disengaged from the extensions in the dialplan. I wouldn't say to make the names meaningless, though; there are different ways to use those names so that they have useful meaning. Just don't make them extension numbers; it's like the TCP/IP boundary between layers. See SIP for an example of the problems such a thing can cause :-) B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime on Asterisk 1.2.24
Tilghman Lesher wrote: I *STRONGLY* recommend that you do NOT use realtime extensions. If you want a dynamic dialplan, the correct way to do it is to segregate your logic and your data (via something like func_odbc), not to stick all of your logic into a database. Should this be taken as a warning to us happy realtime users that it is deprecated, and/or likely to go away? If that's where it's headed (once upon a time a discussion to do it the right way was scheduled for the Atlanta confab last spring, but the topic never emerged there AFAIK) then it ought to be officially deprecated so those of us who use it extensively can begin to plan our migration to other ways of solving the problems that realtime seems (at least to me) to solve nicely. I can deploy large numbers of servers with complex and coherent dialplans with pretty much zero effort on a given new client, and I can also effect system-wide changes across my servers with a single database update. I find it to be a powerful and useful feature. But if there's a better way to do it I'm willing to learn. To my knowledge there are no Postgres ports of ODBC running yet under openWRT, and so I am using the PG module to access my information. At the moment func_odbc wouldn't seem to get the job done for me as per your suggestion above. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4: encryption support
Russell Bryant wrote: Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4.13 and I need to use encryption among Asterisk and my SIP users, and with the RTP data interchanged among users. I prefer the use of ZRTP/SRTP because we use Twinkle and X-Lite/Zfone as our voip clients and they support these encryption mechanism. My question is: do I have to enable any encryption support in Asterisk 1.4.13 ??? Or Asterisk has native encryption support ??? The only VoIP encryption provided in Asterisk 1.4 is IAX2 encryption, which can be used between Asterisk servers. I'm not aware of any IAX2 clients that support encryption. What's the status of SRTP? I remember seeing things floating around about it being under development, but various sotto voce conversations I've had around over the past few days would indicate that it hasn't gained much/any traction. I'd be glad to be disabused of that notion. Thx. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on USB Flash?
shadowym wrote: Size/Speed/write cycles have gone way up, price has gone way down. More common than CompactFlash and no need for an adapter. So is it feasible to run an Asterisk server on something like this? With a MTBF of 1million write cycles coupled with dynamic wear management on a 4Gig USB drive, lifetime is a non-issue. Just wondering how well it works, if it works. My main server for my home and teensy business runs on a Netgear WGT634U running SVN-trunk under openWRT. The Asterisk binary sits on the machine's flash, but all the modules, prompt files, voicemail, etc., goes to the flash. It works just fine. This system doesn't get a huge amount of voicemail, so I don't know about how long it would be before the wear issue would surface (pun?). I know it works just fine. I've been doing this now for almost two years, although with various versions of embedded Linux and various versions of Asterisk. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99bottlesof beer)
Jon Pounder wrote: Quoting John Faubion [EMAIL PROTECTED]: Does anyone know of such a device that I can use over a network? It would be a pain to run a USB cable. I am thinking of devices that are like: I think your missing the key feature of these devices, UPB/X10. UPB and X10 are communication protocols that runs across the electrical wiring in the home. The 1132 box can be programmed using a USB cable to your computer but it doesn't have to remain connected via USB to control the lights and outlets using X10. Additionally you only have to connect one PC-UPB/X10 controller to control the other UPB/X10 devices. You will have to install more devices to make it all work but the commands come across the house wiring. has anyone actually been satisfied with the performance of these powerline signalling devices ? yeah they make a nice cheap demo, but any time I have used them they proved to operate randomly on their own, and not always when they were supposed to. Hardly something I would want for say a door release. Granted in an existing situation there may not be a way to run more wires, but I evaluated them a while back and decided to stay away. I have tons of X10 devices doing all kinds of things, all over. They are in a rural/small-town environment, so things might be different in areas of denser population (caused by other controllers on the mains). For me they work flawlessly. I have light and appliance controllers, as well as X10 thermostats in two houses. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
Andrew Joakimsen wrote: That's horrible. I don't buy too many IP phones these days, but can anyone suggest a place better than the scumbags at VoIP supply? Reading this raised my eyebrows. I'm not sure what the content will do for your readers regarding the company; I am absolutely sure what it does for my opinion of your sense of fair play. Name-calling is the oldest, cheapest trick in the book. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
Matthew Fredrickson wrote: shadowym wrote: Maybe his comments were taken out of context as they don't have the whole interview posted. Why is he talking about queue games, Biologicall and other extremely niche crap when there are huge holes in the basic offering (SLA and SCA)? Considering it is an open source project, anybody that has access to the source code (i.e. everybody) can work on whatever they want to, whether it be SLA, SCA, or queue games for the more light hearted. Amen, brother. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
shadowym wrote: Yes thank you for reminding me it is open source. Thank you for reminding me people can write their own code for it. I'll get right on rewriting the entire sip code. Should only take me a few hours. Including a couple hours to learn how to write c code. How hard can it be! I can't tell whether you're intending to prove the point that was being made, or trying to be sarcastic. Knowing your posting history, I'll assume the latter. But in case you're serious, and you really do believe the coders owe you something, here's another translation of the situation: If you code, if you contribute to the coding effort by intense testing and/or filing bug reports, if you carry Red Bull to the programmers during hacking sessions, etc., then--in the vernacular of the Church of the Subgenius--you buy slack. And once you have slack, you can say, Let's do this, or Let's do that, and the developers will consider it and--maybe--implement it. When, instead, you are 100% slack-free and have been noted before nipping nasty mots at the hands that feed you code, the chances of having your tart remarks about SLA taken seriously are pretty slim. But, and here's the point: It's Open Source. If the developers look the other way when you ask for something, if they don't answer your emails, if they don't drop everything when you demand something and do what you want, FORK IT! Take the code THEY they wrote and do with it what you will. It's free. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
Matt wrote: Absolutely not! However, if someone gave me a free flight, but the plane went down 3 out of the 5 times it took off, yes I would :) Then, if the makes of the plane released a new version where they fixed the problem, but now instead of going down because the motors shut off, it would go down 3 out of 5 times because the rudder would get stuck in an odd position (note this didn't happen before the motor 'bug' fix). Finally, 4 versions later we have a stable plane that can fly for about 2 months before it crashes, however, if you land it each night it will run fine. But, alas, terrorists have found a way to get into the plane and hijack it, so the airline releases nothing more than a security upgrade for it. Now, however, we find that after flying for 2 hours, the wheels fall off the plane. Airplanes are built, maintained, and operated under extremely restricted operational conditions. Asterisk, which derives its very name from the notion of ubiquity, is being used by everyone to do everything, and regression testing in that environment is, um, challenging. I run a fairly recent version of SVN-trunk as my primary Asterisk server. Traffic loads are relatively light because I'm a small-bones player, but I have lots of different channel technologies, many ITSPs, and a network of over a dozen cooperating servers running several different versions. It's not a lie to say that my systems *never* go down as long as they're supplied with power--which in rural Indiana is never a sure thing. I agree with Russell's initial assessment; Matt's phrasing, if not his intent, emanated from the land of the troll. . . if for no other reason than the implication that Digium is solely responsible for the development of the product. FWIW. b. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Stephen Bosch wrote: PSTN service still sets the standard. With infrastructure paid for under a gracious guaranteed-profit monopoly by ratepayers, now being used as a weapon to stifle competition from VoIP, cable, and other emerging technologies. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Stephen Bosch wrote: Douglas Garstang wrote: I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Don't be confused. The answer to most of your questions is no. I don't think he's really confused. Doug has a penchant for the provocative, historically. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk is not sip proxy
Alex Balashov wrote: The distinction between a back-to-back user agent and a proxy is a rather formal one; (. . . . ) I suggest the essence of this mail be distilled and put into a FAQ somewhere, if it isn't already. Thanks. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID Spoofing to be banned in the USA
J. Oquendo wrote: Stephen Bosch wrote: I ask that you treat people respectfully on the list. The poster has a valid point and does not deserve that kind of response. It's possible to disagree and still be civil, and I've no doubt you're able to do it. Thanks, Right sorry list for living in a place called reality. In case (as it seems) you're tone deaf about being a dick, there's another example of it right above this sentence. Was not my intention regardless of how it sounded. (honestly... I'm just rather blunt) I read your first couple of posts, and thought, Where does this guy get off? Mr. Know-it-all; rest of us is stupids. Nothing you have said since looks any different. I hope you never have to ask anyone for a favor. You really do seem way too good/smart/clueful for the rest of us mere mortals. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Me Too mails DOSsing the list (was: Re: [asterisk-users] Voip-info.org)
Compnet Bobby wrote: Same in southern cali! Folks, at the risk of sounding mean, once there are a couple of Me Too emails on a given topic, everyone who reads the list is clued in, and we don't really need another hundred or so one- or two-word emails adding to the already-bad load of traffic the list generates. Very early on, we heard from the site maintainer that they were being DOSsed. I still have many hours of list traffic ahead of me to read, and I despair of how many more of them are likely to be just like this one--and I'm not picking on you, Bobby; you just happened to post the 25th one of these I've read today. Thanks. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Outside lines are just not happening...
Stephen Bosch wrote: If you get dumbfounded responses ask to speak to someone in the programming group (unless they are a tiny little phone company, they will have one). If you open a ticket, it usually means they will escalate the problem, even if the agent you are speaking with has no idea what you are talking about. Best to be friendly with them! And what do you do when they say: We have a modern, relatively-new switch for which that sort of feature change is a trivial click on a GUI checkbox. However, we do not have any tariffed requirement to provide disconnect supervision. So we won't do it for you. Goodbye. And then the call ends. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Outside lines are just not happening...
Stephen Bosch wrote: And what do you do when they say: We have a modern, relatively-new switch for which that sort of feature change is a trivial click on a GUI checkbox. However, we do not have any tariffed requirement to provide disconnect supervision. So we won't do it for you. Goodbye. Ask for the supervisor. Don't quit pestering them until you get someone who can help you. DON'T QUIT. You'd be surprised what persistence will get you (that is, more than just disconnect supervision) My phone company is owned by two brothers, and it was one of them who told me that. I'm borked. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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: Now way OT: [asterisk-users] Need some help with a very simple Queestion..
Stephen Bosch wrote: Is Marmite also available in Ontario, or only Out West? As far as I know, Marmite is available all across this land, from sea to sea to sea. Three cheers for Marmite. IMO most Americans have never even *heard* of Marmite, much less tasted it. And it's quite a hoot to watch someone ingest it for the first time. Always causes a surprised look. Someone should write a book about it--or maybe someone already has :-) b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Improving Asterisk's DNS support
Kristian Kielhofner wrote: The intent of my original message was to try to start a discussion on how we can fix a REAL, KNOWN problem with Asterisk to make it better. I'm not sure what the exact problems are (or even the specific symptoms in all cases), but I am willing to offer anything that I can (money, testing, etc) to fix this problem. I realise you don't know that this is a problem. However, your tone is *not* what I am looking for in this discussion. For those who haven't ever been bit by this situation, please believe that it is not only real but devastating. Kristian is exactly right: there needs to be an architectural fix that will stop EVERYTHING in the server from hanging when lookups are done (and then hang) for a given IAX/SIP peer. I'm not programmer enough to do it, but once it's done I will be eternally grateful. Just yesterday I had a situation where our egress circuit was being used temporarily to do a big remote database dump. As a result, DNS queries to the outside world were taking forever to resolve. An internal server I have, which talks to a single DNS-named peer in the outside world, hung tight, repeatedly, while the T1 was compromised, even though none of its calls had anything to do with that peer. It was the core engine that was hung up waiting. The worst of it is when there are Zaptel cards in the box, it doesn't even handle call pickups and the like once the DNS hang has begun. There are various kludgy workarounds, but at the end of the day it does not seem that the server should hang totally when this situation occurs. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] wireless desktop phones
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone. But of course they want IP. Are there any adpaters
that will give me just enough bandwidth to get it done. The computer
network is all wireless so the phones would have all the bandwidth.
Some of the Wifi phones--at least under the relatively stable conditions
I have here--work very reliably.
I have 3 Starcom F1000s, and a) if they don't have to roam and b) they
don't have to connect dynamically to different servers, work just fine.
FYI. YMMV.
B.
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Re: [asterisk-users] Need feedback on vitelity
Mail list wrote: Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . I would like to express an opposite opinion. I have two accounts with them with lots of DIDs. Everything works fine, and they have been very quick to respond to the few issues we had to work through trying to implement, I think, 16 incoming lines in 14 area codes. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] A request for your input.
Stephen Bosch wrote: Bill Hackensack wrote: On 3/22/07, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello P.S The program that I am using is open source, of course ( www.phpsurveyor.org http://www.phpsurveyor.org)! What part of the survey is running Asterisk? Wow, you're friendly! And helpful, too! I thought the same thing, and am glad you sent that message. Thx. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] voip-info.org status update
shadowym wrote: If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. Suggestion: switch to something else!! Why stoop to use something you seem to disdain so fully? Take your complaints elsewhere. Just a reminder: Asterisk is free. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] asterisk on mini-itx
Chris Mason (Lists) wrote: Here's how I do it. Buy complete fanless system flash card ready unit with four ethernet interfaces: http://www.ibt.ca/v2/items/fwa7204/index.html It is very small, in an aluminium extrusion case, very robust. What kind of money are those things? There doesn't seem to be any price information on the website you linked to. Thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] REALTIME and VIEW ENTIRE DIALPLAN
Davis Sylvester III wrote: Is there a way to view the entire dialplan when using Realtime? I use Realtime and MySQL connector. If you mean the contents of .conf-file based merged with whatever the Realtime engine is supplying, I don't think there's a way of seeing both together. But you can use the standard CLI dialplan revelation tools in conjuction with the standard MySQL table listing tools to see everything in two pieces. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] REALTIME and VIEW ENTIRE DIALPLAN
Davis Sylvester III wrote: Brian Capouch wrote: Davis Sylvester III wrote: Is there a way to view the entire dialplan when using Realtime? I use Realtime and MySQL connector. If you mean the contents of .conf-file based merged with whatever the Realtime engine is supplying, I don't think there's a way of seeing both together. But you can use the standard CLI dialplan revelation tools in conjuction with the standard MySQL table listing tools to see everything in two pieces. B. HOw do I see the mysql stuff from the CLI. I know I can do a show dialplan from the CLI to see the .conf files stuff but not aware of how to see the mysql stuff. From the dialplan, you can't. The essence of Relatime (modulo the caching that it does) is that the server *doesn't* keep configuration state that can be gotten with the Realtime engine; it looks it up dynamically. In other words, the proper tool for seeing the part that lives in DB tables is the tool that comes with the DB that extracts that information from the database backend. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Re: Asterisk Realtime
Mike Hammett wrote: [Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:623 mysql_reconnect: MySQL RealTime: Successfully connected to database. [Mar 7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register: Registered Config Engine mysql [Mar 7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register: Registered Config Engine mysql MySQL RealTime driver loaded. res_config_mysql.so = (MySQL RealTime Configuration Driver) All that looks fine. What do you get when you do realtime mysql status? The next areas to look at would be your DB configs, and debug status when you actually try to use one of the entries in your DB. . . I only use it for iaxpeers/users and extensions, so I can't comment much on its use with SIP or voicemail. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Asterisk Realtime
Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I can’t see any signs that it’s working. I followed and double-checked a few different guides around the net, but haven’t been able to figure it out. You don't say which version you're running. I *think* the syntax is the same for both: realtime driver-name status will show you the status. For postgres it's pgsql for driver name (that's what I use). I think the other driver ids are mysql and odbc. If you don't see yourself connected, that's where to start. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Sending SMS
Supa wrote: Try this: http://www.bayhamsystems.com/asterisk.html Works for me just fine, and it is very easy to get up and running, even with older version 1.2.3 Anyone out there running it against 1.4.0? It built just fine for me, but then it crashes the server when I try to run it. Crashes on both original-flavor 1.4.0 and the newer SVN version. Thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Sending SMSa
Michiel van Baak wrote: On 17:53, Sun 25 Feb 07, Brian Capouch wrote: Supa wrote: Try this: http://www.bayhamsystems.com/asterisk.html Works for me just fine, and it is very easy to get up and running, even with older version 1.2.3 Anyone out there running it against 1.4.0? It built just fine for me, but then it crashes the server when I try to run it. Crashes on both 1.4.x and the SVN-trunk versions. Thx. I assume that you want the asterisk module here. I never tried that, the agi is easier :) Try the agi version, that should work on all versions of asterisk. I solved it. It appears that Asterisk (1.4.x and SVN-trunk) will go ahead and happily build any app_whatever.c code one puts into the apps directory, so getting the file for app_fastSMS and putting it in there causes it to be built. But without first going through the menuselect process before it's built, the appropriate libraries don't get linked in to the module, and after Asterisk loads it very, very funky things begin to happen. This application requires the same libraries as func_curl. There are two fixes--the correct one is to add the appropriate skeleton code into the menuselect-tree source files, which I leave as an exercise for the reader. The kludge is to just pretend that you did, and add the line: MENUSELECT_DEPENDS_app_fastsms=CURL to menuselect.makedeps at the toplevel directory. The INSTALL document that comes with the source code doesn't mention this requisite; I wonder how many people running 1.4.x have actually tried to build it. . . I'm cross-posting to -dev for archival reasons, as an FYI to folks there who might be interested. Hope that's OK with you -dev types. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] CWI, call-limit and incominglimit
Pavel Jezek wrote: it can be also usefull to use 'limitonpeer' option in sip.conf with this, it would not be needed to define separate type=user and type=peer for each phone, instead define one type=friend and apply limitonpeers=yes ;limitonpeers=no; Apply all call limits (limit=) only to peers, never ; to users. This improves handling of call limits ; and device states in certain situations. The user part ; of a type=friend will still be affected by the call ; limit, but Asterisk will only use one object for ; counting the simultaneous calls. I'm a little confused about the comments shown above, which I assume are from sip.conf. limitonpeers=yes would seem to imply that the limit= value would only apply to the peer portion of the sip user. But the included comments say, The user part of a type=friend call will still be affected by the call limit Those seem to be in conflict, but perhaps it's just my parser :-) Could someone clueful explain? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] cisco sip firmware update for cisco 7970
Tim Connolly wrote: You can buy smartnet on a single phone for something like $8 a year. This will get you in legally. Any idea about how specifically to get such a contract? It is rumored to be pretty tricky. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Problem with busydetect and cell phones
Stephen Bosch wrote: Have you tried calling ATT and asking for call disconnect supervision? I realise that this can be a thankless and tedious endeavour, but it IS worth trying. There are almost no commercial switches that don't support this; it's a matter of activating it for the specific circuit in software. Particularly if you have a business line -- you can demand it. All PBXs need it if they use analog lines (and plenty still do) so I'm sure this is not an alien concept to ATT. It's just a matter of finding the right Earthling there who can help you. This might be one of those times where a beer with the technician will get you some joy, if calling Repair doesn't give you any joy. Better luck with ATT than I had with the Monon Telephone Company. They have a switch that's fairly new, so I called them--I'm a loyal but tightly captive customer of the last 25 years. Their chief technician told me, Sure, our switch is new. There's nothing to it more than a setting on a software screen. But we don't have to do it because it's not in our tariffs. So forget it. And then he hung up on me. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Problem with busydetect and cell phones
Stephen Bosch wrote: And then he hung up on me. ...wow. This society is doomed. Actually, it isn't so much society as the legacy telcos. But unfortunately, they've been pretty smart about using the billions that they've stolen from us over the years: they use a lot of it to line the pockets of our legislators, and then have them write laws (such as the recent SBC Benficiency Law in Indiana) that stifle competition in the local loop and put their competitors at a disadvantage. Martin at the FCC has been a disaster for competition; SBC now has all the old ATT properties, and they're just a few new regulatory laws away from having their monopoly back. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] PRI Call Start
Stephen Bosch wrote: And use a different Wiki engine! Augh! (Mediawiki, anyone?) Who runs voip-info.org? I'll bet if you volunteered to take it over, the folks who run it would gladly let you have it And I'd further bet they'd gladly let you run whichever Wiki software you want!! Otherwise, it strikes me as unseemly for you to criticize the way it's being done. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] List problem handling HTML E-mails?
Yuan Liu wrote: My multiple postings to this list this morning got garbled in http://lists.digium.com/pipermail/asterisk-users/, and don't come back from list. (e.g., http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I thought it was Hotmail, so I saved one outgoing mail and checked that it's correct. Anyone else experiencing same? The postings that I saw from you came in HTML format. This one did not. I don't know if that's germane, but I wanted to mention it. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] realtime extensions, labels
Julian Lyndon-Smith wrote: I cannot seem to find any reference to labels in realtime extensions - using 1.4. I've googled until my eyes have bled, and also scoured voip-info.org. Is there anything that helps me here ? You have to have numbered priorities with realtime. This is because (as I understand it, corrections very welcome) in the case of file-based configuration it is parsed once into an internal structure, so the parser can make use of the n construct to autonumber. Realtime is totally dynamic, so that convenience doesn't apply. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)
Jon Pounder wrote: Buzzwrong answer! Don't answer on things you have no idea. and stop providing bad information. you should take your own advice - an acre is 200ft x 200ft - what idiot would pay a consultant $7000 to tell them they need one access point in the middle. This is getting ridiculous. As a farmer's son, I must report that your acre is a bit too small. Perhaps you meant *approximately* 200x200? A square acre is 208.710325571 feet on a side, or 43,560 square feet. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Some queries on g729 license.
Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. I know you didn't ask me, but I think you should talk less and listen more. You have now had two sessions of knocking over the china on this list in the past week or so, and neither time have you come out of it looking very good to other members of the community. As long as patent laws apply in the US--and there is still no looming relief for the idiotic state that things are in--if you use G729 in the US in anything other than extremely restricted uses, you must pay the appropriate license fees. Pretty much period. The topic of free G729 is very shopworn on this list, and if you had spent some time reading past posts before starting in, it would have been a lot better for all of us. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] How accurate is show translation?
Steve Underwood wrote: Paul Hales wrote: When you built Asterisk, it must have refused to build the ilbc codec - I have never seen an Asterisk box that could not transcode ilbc, in over 3 years of working with Asterisk. Most versions of embedded Asterisk will choke unto unusable if they are faced with transcoding t/f iLBC. Without an FPU they can't keep up with the audio stream. Nice while tripping on acid; otherwise of no use whatsoever. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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?
Folks, with all due respect: this thread is now wy off topic, as it has nothing to do with Asterisk whatsoever. Please take it offline, or to ~biz. thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] How to park calls on a specific extension
Lacy Moore - Aspendora wrote: I think that's the problem with the Asterisk community right now. Anytime something is suggested, the response is either write it yourself or deal with what is there. Do you have experience with other big, complex Open Source projects? Do you know of any where the end users sit and pitch requests, and the developers read them and say, Yes, sir. I'm here to serve you!! Does Digium want feedback on what actual, real users want, or not? If not, fine. I won't be making another suggestion. Everyone's suggestions are seen and considered. But what goes into the code is unlikely to be what a user-only has suggested unless one of the developers becomes convinced that a given feature is worth the time that will be required to implement and test it. I think it's childish to tell someone who is requesting a feature to write it themselves. Did you ever stop to think that if they could, they wouldn't be asking for that feature? . . short litany of complaints elided . . . I won't be doing another Asterisk install for a while. Customer #2 has made sure of that by telling everyone how their new phone system sucks. But seriously, the attitude of either write it yourself or deal with it won't cut it for business users. If Asterisk is only for geeks, then fine, it will work perfectly. Have you looked into Asterisk Business Edition? If your customers are having that bad of a time, you may have sold the system before becoming familiar enough with what Asterisk does and how it does it. It doesn't help anyone to have someone do an install and then have the end user thinking the system sucks. If it sucked as bad as you portray it in this mail, the lists wouldn't be so laden with mails--including suggestions from endusers such as you. Complaints are always considered, but calling the developers childish and repeating that complaint over and over in an email isn't likely to do much to advance the cause you've taken on. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
[asterisk-users] Anyone got a dialplan for SPA ATAs for ISN?
After John Todd's talk at Astricon about the ISN project, I spent much of the weekend playing around with it. I have discovered that the default dialplans on my Sipura gear, as well as my Grandstream phones, intercept the * key that is a required part of ISN numbers and interpret it as a metacharacter. Googling for a while has turned up evidence that this can be corrected by a carefully-crafted dialplan for the Sipuras, at least, but the avaialable documentation is, let's say, a little convoluted. I'm wondering if anyone on the list has cracked this, and would be willing to share the gobbledygook string needed to effect the proper behavior. Thanks. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] No Authority Found
Andrew Joakimsen wrote: Another thing is my understanding of the peer, user and friend. I thought that a peer can only receive calls from either a user or a friend, a user sends calls to a peer or friend and a friend is both a peer and a user, however in my production machine I have the following configured: You have it backwards. Peers terminate calls--that is they provide a connection to a remote endpoint. Users originate calls into your system from the outside world. Friends do both, but last I knew the cheeses have deprecated the use of friend in favor of explicit peers and users. And yes, this certainly could impact your connections both inbound and outgoing, causing the error message you report. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Embedded Asterisk
Kristian Kielhofner wrote: Administrator TOOTAI wrote: Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Cory, OpenWRT -running on Linksys WRT- has asterisk packages. [EMAIL PROTECTED]:~# ipkg list | grep asterisk asterisk - 1.0.10-1 - An open source PBX asterisk-chan-mgcp - 1.0.10-1 - a Media Gateway Control Protocol implementation for Asterisk asterisk-chan-skinny - 1.0.10-1 - a Skinny Client Control Protocol implementation for Asterisk asterisk-codec-ilbc - 1.0.10-1 - an Internet Low Bitrate Codec (ILBC) Translator for Asterisk asterisk-codec-lpc10 - 1.0.10-1 - an LPC10 (Linear Predictor Code) 2.4kbps Voice Coder for Asterisk asterisk-codec-speex - 1.0.10-1 - a Speex/PCM16 Codec Translator for Asterisk asterisk-mini - 1.0.10-1 - A minimal open source PBX asterisk-mysql - 1.0.10-1 - MySQL modules for Asterisk asterisk-pbx-dundi - 1.0.10-1 - Distributed Universal Number Discovery (DUNDi) support for Asterisk asterisk-pgsql - 1.0.10-1 - PostgreSQL modules for Asterisk asterisk-res-agi - 1.0.10-1 - Asterisk Gateway Interface module asterisk-sounds - 1.0.10-1 - a sounds collection for Asterisk asterisk-voicemail - 1.0.10-1 - VoiceMail related modules for Asterisk Daniel, Those are ancient! Capouch has MUCH newer packages. -- Mine probably aren't managed as well. I'm a one-person operation with too many irons in the fire!! Has anybody out there, on non-FPU embedded platorms, made any good use of things like ilbc and Speex? I downloaded those packages a while back but they were dramatically unusuable on either the WGT or the WRT models. Maybe I'm missing something? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
[asterisk-users] Netgear WGT Flash-fest at Astricon
Just an FYI to anyone out there who will be attending Astricon and who would like to play around with embedded Asterisk on the Netgear WGT634U platform. If you want to bring your own to the show, I'll be bringing all the appropriate stuff to flash them there with my latest openWGT/Asterisk build. They are available from www.justdeals.com, refurbs, for $44.95 delivered. You'll also need a USB flash drive. I use 256MB, but Asterisk can be set up to use as little as 32MB. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Netgear WGT Flash-fest at Astricon
Kristian Kielhofner wrote: Do you have the necessary components for a serial cable for these little guys? I would like to play with the loader and get a serial console... If you don't have one perhaps we can work on getting the parts before then. I have one, and also will be bringing the necessary components to build another one--although I would prefer not to do so just right now--so that anyone interested can see how the commercial one is constructed on the inside. I have a beginning programming student who was charged today to breadboard up a few more of these, and, if he gets good at it, to make up a few of them. We're using these things like crazy now in quite a variety of use cases; I recently added, at a customer's request, policy routing and traffic shaping capabilities. They have proven to be quite reliable also acting as client WISP CPE. A $90 pair of them provides client CPE, a local premises AP, and a small-volume Asterisk server. I built 1.2.13 for it tonight. Kenny from Digium, a former student of mine, has implemented a proof of concept remote training server on one, where we use screen to provide the student with a way to watch an expert administrator fiddle with a system's configuration. They're great fun, and a harbinger of a future direction in bottom feeder telephony, which is the space I inhabit :-) B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Reception Console
Scott Higginbotham wrote: I'm interesting in testing this. OFF LIST PLEASE, FOLKS!! The list has enough traffic without the 10,000 me too mails that are likely to follow if nobody points out that it's bad netiquette. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] VoipSupply? [Semi-Urgent]
Jay R. Ashworth wrote: On Mon, Oct 16, 2006 at 10:21:29PM -0400, C F wrote: On 10/16/06, VaibhaV Sharma [EMAIL PROTECTED] wrote: I don't think this is a problem because of the snow storm. Yo, all; please take this thread where it belongs, which is the -biz list. It is not relevant to the operation and/or configuration of Asterisk. Please? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Why is this happening?
Matt wrote: Ok I understand all that... Just wanted to confirm that A) it was the remote router mangeling the port and B) that it wouldn't cause an issue (I wasn't 100% sure if it would.. since only the 4569 port is open on the firewall). Could this cause an issue? If only 4569 is open on the firewall, and IAX tries to setup the connection and then move to a port that isn't opened wouldn't this cause one-way audio, or no audio at all? If a remote NAT router has mangled the port that Asterisk is using to some other value, then that is the port--from the perspective of the local Asterisk server--that it will be using to effect communications with the remote endpoint. At the other end, the NAT router will handle the appropriate translation from that port to the port being used on the Asterisk server. I do this all the time, and the reason I say the port being used on the Asterisk server is because IAX is able to handle any number of such mappings without problem. The only problem I have is NAT routers that have very short timeouts set on such mappings. There are ways to get around that problem, but that discussion isn't germane here unless you have identified that as being the situation in this case. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] How big is *your* dialplan??
Douglas Garstang wrote: I don't get it. The clients are ok with their phone systems being down anywhere from minutes to hours? Try googling for cost benefit. I got 135 million hits. Your brain has some very odd twists in the way it works. Or you're a troll. B. ___ --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] RE: Welcome to the asterisk-users mailing list
Issac Simchayof wrote: Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine on the same box. Any ideas? Yes. Don't use such a useless subject for your queries to the list, and you might find them better received. . . The archives of this list is a valuable resource for those doing due diligence before bothering list members. A subject like yours hides the intent and content of your message totally, making it worthless as a subject search target. Why not SIP 2.01 on Polycom? Too late now, though :-) B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Re: Real-time and priority n
Tony Mountifield wrote: In article [EMAIL PROTECTED], Ronald Wiplinger [EMAIL PROTECTED] wrote: Is it exclusive? Either Realtime or priority n ??? If so, what is the better way? I believe 'n' is just a shorthand way of writing previous line + 1, and gets converted into an actual number as the dialplan is compiled. After compilation, the information about whether a line had been given as 'n' or as a specific number has been lost, as far as I know. Rows can be added to a database table at any time. Imagine a series of priorities added to a table using nothing more than n as a priority number beyond the first one. Now imagine wanting to add a new priority in between any two arbitrary entries in the table. How would you even specify which two lines should surround it, when they have no identifying serial number associated with them? Unless you were to add a new field, e.g. priority location identifier, or somesuch. Which does nothing more than move back to the present situation. The extensions.conf parser adds a real priority to each line, but in Realtime that responsibility falls on the DB maintainer. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
[asterisk-users] New ptlib dependency-requirement in SVN-trunk?
Suddenly I can't get SVN-trunk to build anymore; the configure script is looking for something related to ptlib I don't have: checking for /root/pwlib/include/ptlib.h... no checking for /usr/local/include/ptlib.h... yes checking for ptlib-config... no checking for ptlib-config... no Cannot find ptlib-config - please install and try again Starting ./configure --without-ptlib does no good. I had never even heard of ptlib; the header file it found says it's a Portable Windows Library. Anyone with a clue on this I'd be grateful to get things to build again. Thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] SER with multiple asterisk deployment
Douglas Garstang wrote: If your referring to using AVP operations to peek into the SIP message, and determine state, good luck finding documentation on that! From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Douglas Garstang wrote: It won't work, unless you make sure that transfers go through the same asterisk server as the orignal call went through. Using the SER dispatcher won't fix that. ONCE again, design your system correctly and it won't matter which Asterisk box processes your calls - including transfers. No, I won't elaborate, so don't ask. That's funny, Doug, you giving advice to Jeremy. My favorite CS teacher in college, with almost maddening frequency, would answer our questions about the operational characteristics of the software we were working on by tugging ponderously on his chin, looking up at the ceiling, then busting a big grin and then saying, I don't know. What does the source code say? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Re: Very high ping times from 7960 phones
Michiel van Baak wrote: On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have the same here. All between 150 and 250 ms. The phones do work perfectly, only the time in sip show peers is higher then any other phone/device. That is a classic (and, AFAIK innocuous) behavior of the original Cisco ATA-186 ATAs as well. Nobody was ever able to explain why they are that way, but it seems to normal behavior. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] RE: 1.4 Beta 2 Config Problem
Andres wrote:
Andres wrote:
exten = 8600,13,GotoIf($[${CALLERID(number)} = 2013]?50:51)
exten = 8600,50,Set(CALLERID(number)=2000)
You are missing a parenthesis.
It should read:
exten = 8600,50,Set(CALLERID(number)=2000))
I just tried with a fairly recent version of SVN head--a close precursor
of the 1.4 beta code--and the following syntax works just fine:
exten = 99,1,Set(CALLERID(num)=2025452432)
exten = 99,n,Dial(SIP/btel)
I sent this call to a Budgetone 101 that speaks out the CallerID, and it
read back the correct value.
HTH.
B.
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Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote: Hi, Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE, India Pakistan. This is a -biz question, not -users. Also, do you realize how bad it makes you look that you can't even bother to put a subject on your mail? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Digium GUI?
Douglas Garstang wrote: I wonder if the look and feel of this GUI will be completely configurable. If it's not, then I really don't think that's very useful. Service providers wouldn't be able to use it to let their customers manage their own settings, and customers wouldn't want to use it if it wasn't branded with their company info. This might set a record beating out your many prior incidents of tacky behavior. Not content to let Digium release the product and *then* criticize it, you're already getting your licks in before you've even seen it. Audacious! B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Realtime Extensions -- Comments?
Douglas Garstang wrote: It doesn't matter where you turn in Asterisk, there's gotcha's. For example, you can't put the hint stuff into realtime, and there's no inherint way to comment extensions. It doesn't seem like Asterisk is good enough for you Doug. Switch to one of the competitors' products. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Realtime and labels
Douglas Garstang wrote: Does anyone know if realtime extensions support the use of labels? I don't believe so. As I understand it, the dialplan parser internally converts n-type and labeled priorities to a straight numeric format, which is then used internally. Becuase the Realtime engine bypasses that parser, it has to have extensions in strict, old-style numeric priority order. If this isn't correct I'm sure someone will point it out. B. ___ --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] Manager Interface API's
Douglas Garstang wrote: Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. Have you taken your be nice on the lists pill today? The most likely explanation is that people have written these interfaces primarily for their own use, and when they decided to share with others, only had/made time to minimally document them. Do you understand that? You've got me doubting you can't understand such things, so I wonder why you *say* you don't understand. Unless you enjoy being a troll. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Manager Interface API's
John Novack wrote: I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough information for many others to make effective use of their creation How many have struggled through the years with uncommented or poorly commented code when the original creator is off to greener pastures? I have struggled like that on a great number of occasions, and know perfectly what you are describing. But I don't think it's fair to blame people in the Open Source community for not doing pro-grade documentation. They give away what they write; if it's useful, all good. If not, then buy a commercial product, or move to another OS product that has better documentation. Especially in this case, where the overwhelming likelihood is that the programmers wrote the APIs primarily for their own use, I don't think it's fair to be casting Garstangian aspersions. Those APIs aren't big public projects, but rather labors of love that don't have the kind of support staff to handle a robust public face. MO. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Ever donate Software to Digium? If you did your a fool.
EVERYONE PLEASE DON'T FEED THE TROLL!! That post was done only for the sake of generating responses, and we do no one any favors by taking the bait. thx. B. ___ --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] Garbled initial voicemail prompt
Joshua Colp wrote: - Original Message - From: Frank Tarczynski [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thu, 03 Aug 2006 07:16:15 -0300 Subject: [asterisk-users] Garbled initial voicemail prompt I'm having a problem where the very first words of the Asterisk voicemail system prompt are distorted into a loud ear-splitting beep. When I dial my VoiceMailMain extension I get this loud beep followed by the rest of the initial voicemail system prompt. After that everything works fine. I've have this problem under both v1.2.6 (self-compiled) and now under 1.2.10 (under Astlinux). My handset is connected to my asterisk box through an iaxy. With the exception of this voicemail prompt problem everything else seems to work fine. The relevant portions of my voicemail.conf and extensions.conf are list below: If you do an Answer and then a Wait(2) before going to VoiceMailMain in your dialplan does this solve the issue? It might just allow time for everything to settle but I can't say I've ever heard of someone getting audio like you're describing. Back in the day such a thing used to happen as part of the ADSI stuff that was in the codebase. Or at least that is what I was told caused the screeching at the beginning of the voicemail mail entry. This has been a long time ago, but the description in the email describes that behavior perfectly. FWIW. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] RDNIS and IAX2
Douglas Garstang wrote: I'll probably get blasted for this. I hope I'm wrong, and then a little blasting is ok. It appears that Asterisk may have let us down again as a 'carrier grade' solution. Did the list software screw up, or did you post this exact same mail yesterday? B. ___ --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
[asterisk-users] Missing close quote in CallerID breaks SIP. . .workaround?
I posted about this some while back, and at that point was told the remote end is broken, nothing we can do about it. The problem: for whatever reason, some CallerID names come in broken. There is an example CLI trace shown below. My question: is there anything I can do to fix this, since there's nothing I can really do about the broken value being passed in? The call never completes in this instance. . . Thanks. B. ** snip ** Jul 24 01:02:10 WARNING[180]: chan_sip.c:1559 get_in_brackets: No closing quote found in 'Lubbock T sip:[EMAIL PROTECTED];tag=f6ae058c3893f37fo1' Jul 24 01:02:10 NOTICE[180]: chan_sip.c:7112 check_user_full: From address missing 'sip:', using it anyway Jul 24 01:02:10 WARNING[180]: chan_sip.c:1559 get_in_brackets: No closing quote found in 'Lubbock T sip:[EMAIL PROTECTED];tag=f6ae058c3893f37fo1' Jul 24 01:02:10 WARNING[180]: chan_sip.c:6650 get_destination: Huh? Not a SIP header (Lubbock T sip:[EMAIL PROTECTED];tag=f6ae058c3893f37fo1)? Jul 24 01:02:30 WARNING[180]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 101 (Critical Response) Jul 24 01:02:45 WARNING[180]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 101 (Critical Response) ___ --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] Re: asterisk-users Digest, Vol 24, Issue 116
Nobody is going to pay much attention to your help requests if you can't even figure out how to set it up so the subject header reflects what the problem is all about. Why don't you try again with an appropriate subject? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Sipura ATA's Forwarding PSTN Calls to Asterisk
Douglas Garstang wrote: I'm working with a Sipura 3000 ATA here. I'm trying to get incoming PSTN calls on the FXO port to go automatically to Asterisk. I have it working, but I had to configure the ATA to register with Asterisk, which means that all calls are being sent to Asterisk with a caller id of the username used to register with Asterisk. I want the real caller ID to be sent to Asterisk, which means I don't want the ATA to register. The badly written Sipura docs aren't clear about how to do this. Anyone set this up? That's not correct. My SPA-3000 FXO port registers with my Asterisk server, and when the PSTN calls come in, it uses the incoming caller's CallerID for the call. Sounds like you have something misconfigured. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Codec Negotiation
Douglas Garstang wrote: Can't put it in a realtime database. We have multiple Asterisk boxes in a cluster, and it's a well known fact that multiple Asterisk boxes using realime cannot query a common MySQL database. Sounds crazy, but true. You spread some amazing well-known facts on this list. As usual salted with your typical choice of words that implies that Asterisk has crazy flaws that no sane programmer would countenance. I have a dozen or more Asterisk boxes that all query the exact same Realtime database. The setup works fine, and the time to deploy a new station with very elaborate functionality is reduced to minutes. The ability to rearrange behaviors on the fly is also a great feature. I love ARA. I use Postgres and not MySQL, but I can't believe that the choice is SQL engine would make a difference. I think you confuse the requirements of your deployment scenario, which a few minutes ago on this list you yourself characterized as ridiculous, with underlying common features of Asterisk used in quotidian circumstances. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Codec Negotiation
Douglas Garstang wrote: Would you like me to dig up the posts from Keving Fleming stating that this is known not to work Brian? As I recall those posts have to do with the way your particular setup required ARA to work with a failover/redundant cluster system you were building. Beyond that I'm not really interested in getting into a pissing contest. I have ONE SQL table called extensions_table on ONE SQL server, but have maybe 20 SIP phones using that same database, placing calls from 10-12 separate Asterisk instances. I was calling into question your presenting a well-known fact that appears to be incorrect. If Kevin sees this and wants to chime in to support your statement and tell me that my experience is somehow an illusion, he's certainly welcome to do so. I have experienced the taste of crow, and eat it when needed. You? Can certain situations be construed where ARA will not do exactly what the administrator wants? Apparently, from reading some of your posts, true. Can multiple Asterisk servers be set up to use a single database instance to store common configuration information? Certainly true, from my and many other people's experiences. The thrust of my post was to refute the fact, and to suggest you perhaps adopt a little less inflammatory rhetoric when you post to this list. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Sipura ATA's Forwarding PSTN Calls to Asterisk
Douglas Garstang wrote: Here's my invite Brian. The From: is always going to contain the auth id the ATA used to register with Asterisk. INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK208dd2be;rport From: Cody XXX-527-7107 sip:[EMAIL PROTECTED];tag=as3a94778b To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] And here's one from a call I just placed. Note the dissimilarities between the From: and Contact: fields on mine and the snippet of yours shown above. I suspect there is an option somewhere on one of the PSTN tabs on the SPA-3000 that has to be set correctly to enable the pass-through. I don't have time right now to play around with it--my system is working just fine :-) 192.168.1.1 is my Asterisk server, and the ATA is at 192.168.1.113. AstIn is the display name I chose for the registration, btw. B. INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.113:5061;branch=z9hG4bK-3c04a2ec From: Capouch B sip:[EMAIL PROTECTED];tag=5e2ab9e072a1a2cco1 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: AstIn sip:[EMAIL PROTECTED]:5061 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Please suggest me Best VoIP Service Provider
Take this to the -biz list, PLEASE. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
[asterisk-users] Using dproxy to solve no DNS hangs everything problem?
The subject pretty much says it. Wondering if there's anyone out there who, as an alternative to hard-coding IP addresses in /etc/hosts, has implemented dproxy or somesuch to enable Asterisk to survive DNS outages. I had a royal mess on my hands this morning after my Internet connection went down for a while. No DNS = No Anything if there are FQDNs in the conf files. The server hangs, affecting the Zap stuff too, which is WAY bad for me. I need to bite this particular bullet. My POTS line was offhook all morning because my Internet was out, and the first call that came in caused everything to freeze up and the channels were all hung. Thanks. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Using dproxy to solve no DNS hangs everythingproblem?
James Harper wrote: This is a problem that affects more than just asterisk, so I'm sure there are solutions out there! One thing you might do is to put a trailing '.' on all fully qualified DNS names. Without the '.', the system will first try appending the default domain(s), which will cause extra lookups and delays. If all the names that you care about can be resolved locally without needing the internet then the trailing '.' should make it work much faster. Of course, I'm assuming that you do have a local DNS of some sort, and the delays you are seeing are caused by your systems thinking they want to look up external names. Well after three hours of playing, and knowing it's rank awful to respond to your own posts, I would like to review what I've learned and maybe save someone else a few million headaches. I installed dproxy, which is a caching nameserver, and thought at first my problems were all solved. I could lookup names just fine on the machine, I saw the names show up in the dproxy cache file (including the names of the IAX and SIP servers I register with). I made a test call through one of my ITSPs and it worked just fine. Then the sky fell: a PSTN call came in, and as the server rang the phones in the house (one of which is a SIP phone) things locked up tight, just like there wasn't any DNS!! I did a little boinking around on the CLI, and noticed that all my IAX registrations had gone just fine, BUT NOT ONE OF THE SIP REGISTRATIONS succeeded. I'll cut to the quick, although I have to disclaim this as the best guess I have: it turns out that if the DNS server the Asterisk box is pointing to doesn't do SRV lookups, and srvlookup=yes in sip.conf, it's Goodbye Joe. I set that option to no and restarted. Now everything appears to be working just fine. Whew. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Using dproxy to solve no DNS hangs everythingproblem?
C F wrote: Thanks Brian for your work, I have had the same problem I installed dnsmasq and I *think* the problem is gone now, I'm repeating I think, I'll only know when the internet goes down again. Your post inspired me to do one more little test: I removed the default route from my Asterisk server, waited until the error messages started streaming out of the CLI, and then called my POTS line. YMMV, but for me it worked like a charm. If this actually works on an ongoing basis, it will be a great relief to me and to many of my customers. Of course, for now that means no SRV. For me, that's a small price to pay :-) B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] app_conference DTMFs?
Henry J. Cobb wrote: I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. I hate to me-too, but my experience was identical. Crash after crash, and I tried everything that was suggested (limiting codecs, primarily). Something is weird there in that for some it appears to work perfectly, for others not at all. . . FWIW. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Re: for you guys setting up customer offices...
Benny Amorsen wrote: BC == Brian Capouch [EMAIL PROTECTED] writes: BC Everyone's mileage varies, and IMO it doesn't do any of us any BC good for negative opinions to be presented to the public as fact. BC You disclaimed, indeed, but you would have been better off to say BC something like, Grandstreams have been problematic for me in my BC application space. Perhaps that would be more diplomatic, but the truth is that Grandstreams really are junk. Sound quality is bad both ways, tolerance of packet loss and jitter is nil, and the user has to be willing to reboot the phone once in a month. I hate to prolong the argument, but I respectfully disagree. I've over a dozen of them in the field, most of which are in light/medium use on a daily basis. Generally the only reboots they undergo is when our notoriously flaky rural electric power fails. I have a business partner who has even more of them than I do, and he is equally satisfied with their performance in his industrial shop type application. The talking CallerID function is a majorly good feature, and I keep one of them in my office just so I'll know who's calling without having to go look at the phone. I wish you bashers would lighten up a little bit, and not treat those of us who are satisfied with phones as if we were too stupid to know what we can live with. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] for you guys setting up customer offices...
calvis wrote: Grandstreams are junk. (I have had bad experiences with them) The former doesn't necessarily derive from the latter :-) Others of us have found them to be an excellent low-cost solution that puts VoIP in places it otherwise would not be economical to deploy. Everyone's mileage varies, and IMO it doesn't do any of us any good for negative opinions to be presented to the public as fact. You disclaimed, indeed, but you would have been better off to say something like, Grandstreams have been problematic for me in my application space. Thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
Douglas Garstang wrote: Somewhat off topic... I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying to get all calls forwarded to Asterisk. However, (and this is hard to believe), the docs say that 1-stage calling (I presume that means no PIN is required) is not possible with FXO-VOIP calls. I somehow managed to get it to work on another SPA-3000 once before ... although I don't know how to replicate it now. Has anyone done this? Can you provide any pointers? Thanks. I make and take calls on the FXO port of my SPA-3000 routinely. On the PSTN Line tab of the advanced screen, I have this in the first Dialplan entry: (S0:[EMAIL PROTECTED]) Then below, I choose that dialplan (in my case, 1) for the value of PSTN Caller Default DP Also, of course, you have to set the SIP server, username/pw, etc. I have mine register, and because the FXS port is already on 5060 I use 5061. HTH. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
Thomas Kenyon wrote: For some reason when I do this, It only works if I have callerID switched off, otherwise I get authentication errors. Do you know of anyway to bulk-save the contents of all the config screens on that unit? If so, I could scrub the passwords and send you the config for the one I'm using. I just checked; I am getting the CallerID just fine when I bring calls into my Asterisk box via the SPA3K. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Best VoIP provider for Asterisk
[EMAIL PROTECTED] wrote: Termilink, at www.termilink.net http://www.termilink.net Just out of curiosity, would you happen to be affiliated with this provider? If you get my drift. . . . B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Re: Digium Hardware Reliability
M.Hockings wrote: Mike (totally UNimpressed with Digium) Point taken. I was not so much point fingers but asking what my expectation should be and maybe shedding some frustration. I don't really have a lot of experience with this kind of communications gear All the more reason for you to fully inform yourself *first*, and then start posting negative drivel to a public mailing list. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
[Asterisk-Users] Error in config sample for GoToIf?
My teeth are on edge after this one. A couple of perfectly good hours
of my life, and I still don't know what's going on. . . .
The extensions.conf.sample that comes with the current SVN trunk has
this line, in an example that shows how to use ChanIsAvail:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
I couldn't get this to work unless I surrounded the first part of the
test with quotes, too, like this:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
Leaving aside the completely separate madness of trying to determine
just what values mean what for the variable $AVAILSTATUS (which I would
be glad to receive a pointer to), is it indeed the case that the example
in the distribution is in error, or is there some other subtle rule that
is causing the behavior of this line to be correct with the extra quotes
but incorrect otherwise?
Thanks.
B.
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Re: [Asterisk-Users] Error in config sample for GoToIf?
Brian Capouch wrote:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
I couldn't get this to work unless I surrounded the first part of the
test with quotes, too, like this:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
Ooops.
Actually, I mis-pasted one of my intermediate attempts there that didn't
work. So sorry. My excuse is that I've gone daft. This is the line
that actually seemed to branch correctly (although not with a 1 in the
test, but that's part of another question :-))
exten = s,n,GoToIf($[${AVAILSTATUS} = 1]?autoanswer:fail)
Note the extra $ ahead of the leftmost brace. . .
There are many permutations of braces, dollar signs, and quote marks in
the various examples on the Wiki, btw, many of which note that other
examples are incorrect. . .
B.
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Re: SV: [Asterisk-Users] Error in config sample for GoToIf?
Jon Schøpzinsky wrote:
Hello
As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail)
${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1
Through more testing, the double quotes I used seemed superfluous; if
you use them in both places, or in neither, it works the same.
But your example above lacks the $ ahead of the left brace. It is
*that* which I now believe is in error in the example.
Plus there seems to be confusion, on the Wiki at least, as to what
values mean what for ${AVAILSTATUS}
Thx.
B.
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[Asterisk-Users] Oh oh. Micro$oft just noticed VoIP
It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. Let them talk. What's it hurt the rest of us? We have seen the wages of tortured English sometimes unleashed on the list. If they're getting the job done, I say hit the Delete button and get on with your life. If 80% of the list traffic were in foreign languages, then I would say we would have an issue. MO. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Upgrading old version of Asteriak - changes
Martin Joseph wrote: Huh, I never looked at that file before (phone.conf). Actually they seem to refer to a Linux telephony interface? Anyone please care to elaborate on what the phone.conf file is really for? The wiki just has a copy of the default file... It's the conf file for the phone driver, which also I seem to recall is ixj or something. It interfaces to the Phone Jack and Line Jack telephony cards. Google for them. They've been around forever, IMO they're not very useful. HTH. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Don't use CDRTool From AG-projescts
[EMAIL PROTECTED] wrote: hello to all, I advice you to not use Harry!! Only one post is needed for each of your silly complaints. Please don't give people even more reason to relegate you to their killfiles. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] syntax error
Rob Thomas wrote: And once again I am reminded why I shouldn't bother helping people here. Not even a 'thanks'. Dang, dude. If everyone who got helped on this list responded to the list with a thank-you, list traffic would go up by at least 10% :-) And it's a PIA to manually circumvent the list software's hijacking of the reply address; one has to cut-and-paste. That makes private thank-yous kind of a pain, too. Overall your helpfulness is certainly appreciated. It sucks some of the value out of it when you whine about being explicitly thanked for it, IMO. MO, and nothing more. Thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Anyone using VoIP WiFi phones?
Warren wrote: As this is to be used by company execs when travelling that is not a viable option. The Netgear WIP300-NA looks interesting. Amyone used that one yet? Sorry this is on-topic for the thread but not what you've written up there. I have some Starcom F1000s (802.11b model). They are not stellar, but they're serviceable if they can be placed in a good-signal environment. The mikes aren't that hot; people at the other end complain of muffled sound. But they're certainly intelligible, and so far they're the best of the lot that I've tried, which admittedly is a pretty small subset. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
