Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls
Ron Senykoff [EMAIL PROTECTED] wrote: I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Can you tell me which party this is for, so I can ensure I never vote for them? -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ --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] LiveVoip is Bankrupt
Matt Riddell [EMAIL PROTECTED] wrote: [...] Looks like the Daily Asterisk News will need a new host. So, unless anyone can donate space for a custom php and mysql based site, it will be hosted in either New Zealand or Italy. Offers? How much bandwidth does it consume? -- You fall out of your mother's womb, you crawl across open country under fire, and drop into your grave. - Quentin Crisp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
David John Walsh [EMAIL PROTECTED] wrote: I quite like the idea that came about earlier with regards to Romand and Greek gods, I am thinking (if I ever get off the phone to google today) of findind the roman and greek gods of communication.. You are thinking of Mercury and Hermes, the Roman and Greek names respectively for the same god. You may have heard of Mercury Communications Ltd., so your idea isn't entirely original ;) -- Room Service? Send up a larger room. - Groucho Marx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P delayed ring on incoming calls?
Joseph Gutowski [EMAIL PROTECTED] wrote: [...] I wasn't suggesting Asterisk should magically be able to pick up the call before it rings at all, just that if my old roommate could manage to dive across the room and pick up half way through the first ring 99% of the time, surely a computer could do it (if it wasn't waiting for caller ID or distinctive ring determination). This tangential thread was referring to the UK. Your roommate would have to be very alert and fit to be able to realise the phone is ringing and answer it within 400ms :) With a 20Hz ringing current, that's just eight cycles of AC per ring. A piece of kit that tries to detect ringing without just leaving it to finish ringing is likely to suffer from false positives. And the 1 ring wasn't constant -- sometimes it's one ring, sometimes it's 3 -- with no apparent reason (test server with nothing to do except answer one X100P and play an IVR menu). Well, the X100P (or at least a clone) is an unreliable piece of junk anyway, so this doesn't surprise me. Mine has now taken to randomly answering the line even when there's no inbound call, and has now been relegated to being just a Zaptel timing source. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key Please contribute to the beer fund and a tidier house: http://search.ebay.co.uk/_W0QQfgtpZ1QQfrppZ25QQsassZpndc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P delayed ring on incoming calls?
Joseph Gutowski [EMAIL PROTECTED] wrote: [...] Either way, the best I've ever managed on the X100P's was 1 ring before Asterisk picks up and starts doing its thing. Well, when you think about it, it's hardly going to pick up after zero rings, is it? :) -- Beer is proof that God loves us and wants us to be happy. - Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P delayed ring on incoming calls?
Gavin Hamill [EMAIL PROTECTED] wrote: On Friday 22 April 2005 12:07, Peter Corlett wrote: [...] In the UK it's entirely possible - the CallerID info comes through as encoded data before the first ring has taken place :) Polarity change, a burst of V23 data, then the normal rings A good point, but I gather the X100P can't detect a line inversion, so it's pretty much still got to wait until it sees loads of RICH, CHUNKY VOLTS to know there's an incoming call. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key Please contribute to the beer fund and a tidier house: http://search.ebay.co.uk/_W0QQfgtpZ1QQfrppZ25QQsassZpndc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP to PTSN provider
Chris Hills [EMAIL PROTECTED] wrote: [...] There are some providers who can terminate some, but not all, 1800 numbers for free. (If they could terminate all 1800 numbers for free, then we'd use them!) I don't understand - I thought all 1800 numbers were free? They're not like UK 0800s - the costs to the recipient vary depending on where the caller is calling from, so access can be restricted based on area. If your PoP isn't in the allowed area, you're SOL. -- If at first you don't succeed, failure may be your style. - Quentin Crisp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how can i connect a cost display on asterisk
[EMAIL PROTECTED] wrote: Johannes, I would be curious to know if there is a solution for this. Another solution is that you buy a call meter. Which is a small box that can be placed in front of phone phone and that can display costs. FXS-- call meter -- analog phone This call meter needs to be programmed with a table inside and a rate for each destination. It depends on the type of cost meter. One of BT's products is the Meter Pulsing Facility which sends a short 50Hz longitudinal tone on supervision, and just before a unit has been consumed. BT scrapped unit charging in the mid-90s but this particular bit of legacy remains. It's intended for payphones where you charge, say, 10p for a unit and want to know when the 10p has been consumed. That's why you can sometimes hear a buzz on a BT payphone a few seconds before the credit drops, because the longitudunal pulse sometimes breaks through into the audio path, even though shouldn't. I suspect this is because they payphone isn't properly earthed. A cost meter (or paypgone) that determines cost without exchange assistance will suffer from inaccurate pricing information and an inability to determine the start of supervision. Still, given that BT charge a hefty wedge for the MPF, some people just stick a COCOT on a standard exchange line and hope it's good enough that they don't get ripped off. -- Common sense is the collection of prejudices acquired by age 18. - Albert Einstein ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country/city codes
VoIP Services [EMAIL PROTECTED] wrote: [...] Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 I think you'll find that the country code for Bermuda is not 441. I'd have to find a telephone directory to check, but I bet the country code is actually 1, and 441 is the area code. Does anyone know a formula for determining which part of a dialled number is the country code and city code ? There's no formula - you need to use a look-up table. -- She's the kind of girl who climbed the ladder of success wrong by wrong. - Mae West ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR writing incorrect data to pgsql tables
James Bean [EMAIL PROTECTED] wrote: [...] Every outgoing call regardless of whether or not it is answered or busy or just rings out in the database the entry has the disposition as ANSWERED, instead of BUSY or NOT ANSWERED. As a test I intentionally rang numbers that would be busy or wouldn't be there to answer the call. Anyone got an idea where it might be going wrong? Are you using analogue lines? Such lines are considered answered as soon as the number has been dialled by the Zaptel interface. -- Marriage: a souvenir of love. - Helen Rowland ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR writing incorrect data to pgsql tables
James Bean [EMAIL PROTECTED] wrote: [...] I am sorry I did not see anything in any of the docs about analogue lines causing ANSWERED response on all calls. Could you point me in the right direction to a fix or setup that fixes this situation? The only real fix is to get some form of digital service, either ISDN or VoIP. There is no reliable means to detect when a call has been answered on an analogue line, so Asterisk doesn't bother trying. The usual kludge for analogue PBXes is to assume that a call was answered only if the recorded time is longer than a certain number of seconds. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR writing incorrect data to pgsql tables
James Bean [EMAIL PROTECTED] wrote: [...] Is the kludge done at the software side when the data is pulled out for accounting and being under say 45 seconds is a no answer or busy? Or is there a tweak that can be done at the database itself? Since you're using PostgreSQL, you can use a trigger to mangle the data before it hits the database. In fact, there's no reason why you couldn't log to a view rather than a table (but again, you will need a trigger for the actual INSERT.) For MySQL and other glorified flat-file databases, you would need to postprocess the data. You may feel more confident skipping triggers and doing this anyway. So by that any calls that go out over the net using IAX to the telco are considered digital and will report correctly? Yes. You will probably be able to make the simple assumption that if dstchannel ILIKE 'Zap/%' , you're going to have to fudge it, otherwise it's correctly recorded. -- The intuitive mind is a sacred gift and the rational mind is a faithful servant. We have created a society that honors the servant and has forgotten the gift. - Albert Einstein ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loader.c:301 __load_resource: libpt_linux_x86_r.so.1.8.1: cannot open shared object file... [solution found, but quick question]
Paul Belanger [EMAIL PROTECTED] wrote: [...] My question is, how come the LD_LIBRARY_PATH defined in /etc/profile did not link the libs properly? LD_LIBRARY_PATH is occasionally ignored for security reasons. If you wish to globally add a directory to the library search path, you should put it in /etc/ld.so.conf. You may want to re-run ldconfig afterwards to clean up and correct symlinks. -- The young always have the same problem - how to rebel and conform at the same time. They have now solved this by defying their parents and copying one another. - Quentin Crisp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Swap Memory get used totally
Angel Diaz [EMAIL PROTECTED] wrote: Time to time, my asterisk goes down.Verifying with TOP, I see the swap memory of the computer get used totally but, I don't see what the process is using it. Hereis a copy wath I see doing top. Does somebody have an idea ? I have no idea, because there's no problem with your swap space: 08:49:19 up 5:23, 1 user, load average: 0.50, 0.70, 0.64 35 processes: 33 sleeping, 2 running, 0 zombie, 0 stopped CPU states: 19.4% user 11.2% system 0.0% nice 0.0% iowait 69.4% idle Mem: 222992k av, 191988k used, 31004k free, 0k shrd, 68604k buff 120700k actv, 464k in_d,2384k in_c Swap: 457844k av,1060k used, 456784k free 67764k cached You've just 1MB of your 447MB of swapspace. It looks perfectly normal to me. -- A man will joyfully pay a lawyer five hundred dollars for untying the knot that he begrudged a clergyman fifty dollars for tying. - Helen Rowland ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIDs anywhere but here?
Jay Milk [EMAIL PROTECTED] wrote: [...] I'd prefer to have a number somewhere outside the NANP, preferably an asian country. This number will (obviously) be low-volume (minutes/month at the most), and shouldn't cost more than a couple of bucks. Maybe a list member knows and/or is using one? How about sipgate.co.uk, who dish out numbers for free? For non-UK residents, you can choose your number from: +49 1801 777555 +44 845 004 (Local) +44 870 478 (National) +44 7094 820xxx That's probably in order of likelihood of being reachable from outside the UK. +44 8xx is generally diallable from the civilised bits of the world. +44 70xx may have issues. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect line is busy with Zap?
Ian Chilton [EMAIL PROTECTED] wrote: [...] Is there any way of detecting whether something else is on the line before picking up on this channel? No, but you could insert a privacy adaptor device to prevent the FX100P from attempting to dial out. -- My mother protected me from the world and my father threatened me with it. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Instances of Asterisk
B. J. Bomar [EMAIL PROTECTED] wrote: I have a quick question for the list. For what reason would you have multiple instances of asterisk running on a single box? I can maybe see it if you have multiple IP addresses, but other than that I am drawing a blank. It could be useful if you're testing different versions, or the box has been partitioned in such a way (e.g. with UML) that there are different administrators for the multiple instances. -- The golden rule is that there are no golden rules. - George Bernard Shaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PrivacyManager 10 digit limit.
David J Carter [EMAIL PROTECTED] wrote: I thought the standard for the UK was 11 Digits in length, (save some old 0845, 0800, 0870 numbers), but most of these are transported to normal 11 digit numbers. The UK number plan contains 8, 10 and 11 digit numbers if you count the leading zero access digit as part of the number, as is the convention. More specifically: - 0845 is all 11 digits, except for 0845 46 47. - 0800 is a mix of 10 and 11 digits, except for 0800 . - 0870 is all 11 digits, no exceptions. - 0500 is all 10 digits. - 01xxx is almost all 11 digits, but there are some 10 digit exceptions, e.g. 01527 6. - Everything else (01/02/05/07/08/09) is 11 digits. One thing to note is that the Oftel/Ofcom codelist.zip is not reliable for determining digit length of 0800 numbers. -- Weirdness is ... when you get mail from a guy called Dick Girth, complaining about penis enlargement spam. - Suresh Ramasubramanian in the Monastery ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call files
Gary White (Network Administrator) [EMAIL PROTECTED] wrote: Is there a way to use the call files with the dial plan rather that directly specifying the channel and number? I have a notify application that I would like to get calls form whenever I transfer calls. I don't know if it's the right answer, but the first thing I'd try for this would be the Local/ pseudo-channel. I'd be quite surprised if it didn't work, to be honest. -- Euphemisms are unpleasant truths wearing diplomatic cologne. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] calculating bandwidth on DSL?
Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: when calculating bandwidth requirements on DSL, does the DSL technology used (bridged,PPPoE,PPPoA,whatever) matter? The techonology won't affect bandwidth as much as it affects latency and jitter which are rather more important for VoIP. Will PPPoE include Ethernet headers included in the user's purchased bandwidth? etc.. That's presumably down to what the billing department thinks it can get away with. -- The truth is an ambition which is beyond us. - Sir Peter Ustinov ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
Mark Phillips [EMAIL PROTECTED] wrote: [...] Could some clever wag that deals with the language bits of * create some other languages like British, Aussie, SouthAfrican. I'd also be looking for Welsh too (anyone here speak Taff?) I don't, but I know people who do. I get the distinct impression that Plaid Cymru could put up a candidate in Birmingham and win a seat. How about Georgie (I'm kidding about that one). You may be kidding, but at least when I was calling Newcastle in 1998-ish, Digital Dot (as we affectionately call BT's spoken announcements) had a completely different accent in the 0191 exchange to the rest of the UK. It wasn't a SysX/SysY thing, but apparently special announcements just for that exchange. (I've also heard custom announcements on 01902, but thankfully not in Wulv'rumpt'n dialect.) If I were commissioning a voice, I'd probably go for an educated Scottish, Welsh or Irish accent. The London and Essex voice seems overused and also irritates me. The bonus is that a provincial voice actor should be cheaper than a London-based one. All these modes of English are more than just a dialect. My 7 or so years as an Ex-Pat in the US have taught me that American really is a valid language. Whilst most of us English speakers can cope with American we'd be a bit suprised when calling a VM system in Slough, Cooperpedy or Pretoria only to be spoken to in American. I *think* most people are aware that it's the voicemail system that's American, not the company they're calling. I'm usually more surprised to hear a British speaker in voicemail prompts :) Am I just ranting here or does someone get my point? Well, it'd definitely be nice if all our voice prompts were consistent, but as it is there's an occasionally jarring mix of the Digium lady and a bloke from our office. It works though, and callers don't get confused, which is the main thing. -- I want to know how God created this world. I am not interested in this or that phenomenon, in the spectrum of this or that element. I want to know His thoughts; the rest are details. - Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
Bill Seddon [EMAIL PROTECTED] wrote: My wife has been recording the text published on the wiki. A couple of questions for you: 1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? I'd probably go for please enter the full national number, including the area code. A play with BT's automated services on 150 (I'm thinking particularly of the Friends and Family number change section) should be fruitful as to what BT thinks is the best wording. 2) The recordings seem dull on playback even though we are recording using a good quality microphone with matching impedance. Initially we recorded using 16 bit/8K sampling on the basis that this is what is required by Asterisk but that was really terrible. So we're sampling at higher rates on the basis that we can use sox to change it as necessary. Any thoughts on what we can do to make the recordings sound sharper? How about making the recordings on a telephone? Leave yourself a voicemail containing the choice phrases, then edit them out using something like Audacity. -- I have four children which is not bad considering I'm not a Catholic. - Sir Peter Ustinov ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Crossed lines - a worrying problem.
Nick Barnes [EMAIL PROTECTED] wrote: [...] I just made a call via BT to a mobile. Then an incoming call came in and Ann else answered it - it made my call go completely fuzzy and I could hear what the woman on the other line was saying to Ann but I couldn't hear my conversation! When Ann's call finished - mine went even fuzzier and all I could hear was computer and internet noises. Yeah, I've had that happen too, even on a bog standard POTS call. Has anybody got any ideas what could possibly be causing this worrying problem? I'd blame the mobile network. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] couple basic questions
John Williams [EMAIL PROTECTED] wrote: [...] For example, walmart.com has microtel boxes with no OS. Will RH9 and Asterisk run on these boxes? I can't see why not. The only thing that fouls up Linux on generic cheap PCs is some of the weird video chipsets, but who cares that X won't work? You don't want X on a server anyway. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free MOH MP3
Wiley E. Siler [EMAIL PROTECTED] wrote: [...] Does anyone know where I can get some royalty free, cost free music for my music on hold? The stuff at www.zongoftheweek.com is CC-licensed so should be fair game. Whether you want to inflict some of it on callers is another matter :) -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blocking the 'Do Not Call List
Chris Shaw [EMAIL PROTECTED] wrote: That is a matter of opinion and not in any way factual SQL, just as everything else, is as secure as YOU make it... As you said, it's a language for querying relational databases, it has no knowledge of security. That's what firewalls, encryption and strong passwords are for... However, for the purpose of blocking numbers based on a do-not-call list, it will work perfectly fine. It's lightweight, fast and relatively efficient... You seem to be conflating SQL and MySQL, which is confusing your thoughts and arguments somewhat. You are correct that my opinion that MySQL is a poor quality product that's worth avoiding is just that, opinion. It is however formed over many years as a professional developer having to use and manage other people's MySQL databases and watching how they tend to get corrupted over time, and I just don't trust it any more. I don't think I said that it was a bad idea to put a DNC list in a SQL database, although I did propose a more efficient alternative that might appeal to some people. If you're looking for a good Open Source database, Postgres is rather nice. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: VoIP SPAM, what's next ?
Christoph Rothe [EMAIL PROTECTED] wrote: [...] That's right. Here in Hamburg, Germany one day before our elections my phone rang and there was a recording from one of the big parties that reminded me to vote the right ones ;-( It could of course have been a joe-job by another party... -- Madam, there's no such thing as a tough child - if you parboil them first for seven hours, they always come out tender. - W.C. Fields ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blocking the 'Do Not Call List
mattf [EMAIL PROTECTED] wrote: [...] As for speed, AGI scripts that we use on a daily basis do thousands of searches a day through a 800,000 record table in less than a second(on a dedicated 3.2GHz MySQL DB machine) so looking through a million shouldn't be too bad. Asterisk will wait for the AGI to finish. If less than a second implies best part of a second, then that's a bit slow, although probably still good enough for this application. (But I personally avoid MySQL for security reasons.) If you want *really* fast, create a CDB from the phone number list periodically, and search that. Doing a CDB lookup takes no more than 3 disk seeks, so you're talking 20ms at most for a search. -- Fashion is what you adopt when you don't know who you are. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blocking the 'Do Not Call List
Gabriel Millerd [EMAIL PROTECTED] wrote: [...] If less than a second implies best part of a second, then that's a bit slow, although probably still good enough for this application. (But I personally avoid MySQL for security reasons.) how is SQL insecure? SQL itself is just a standard for querying relational databases, and does not in itself specify or require security, nor does it have inherent security problems. MySQL is a toy database that mostly ignores all but the most superficial of the SQL standard and has a large catalogue of bugs and misfeatures that can cause silent data corruption. -- First they blacklisted the porno spammers and I emailed nobody for I was not a porno spammer. Then they blacklisted the open relays and I emailed nobody for I was not an open relay. Then they blacklisted the ISP dialup subnets and I emailed nobody for I was not on an ISP dialup subnet. Then they blacklisted ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Walter Klomp [EMAIL PROTECTED] wrote: [...] However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. This is normal and expected behaviour, at least for POTS lines I've used. When you receive a call on a POTS line, you can't clear it by just hanging up. On a POTS line from BT, you can force-clear an inbound call by hitting recall/hookflash then hanging up at the dialtone. The phone will ring for a few moments and then clear the call. -- IIRC the USA blew up their international telephone exchange very early in the war. Was that bomb sponsored by ATT or Cisco? - Mark Clayton and Tim Clark showing cynicism is alive and well in uk.telecom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux for Asterisk
Sebastian Nocetti [EMAIL PROTECTED] wrote: All distributions are based on same kernels... And in my opinion, Kernel is who does all work in an operative systemm.. I am wrong?... Sort of. libc is the other thing that can affect performance. However, any distribution worth its salt will provide a selection of optimal kernel and libcs because of this. The effort of building a custom kernel and libc is probably worthwhile, but beyond that you should probably spend your efforts elsewhere other than recompiling stuff for the sake of it. Gentoo's performance improvements from recompiling the world are usually more psychological than practical. -- [About a discussion of heavily customised cars.] I thought they were talking about cheap whores - smelly, ugly, brightly coloured, waste of money, and got a cock inside them most of the time. -- Will Hargrave in uknot ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux for Asterisk
Eric Kirkland [EMAIL PROTECTED] wrote: Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? Best is highly subjective, and asking is likely to provoke a holy war ;) I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. Yes, I found that the TTS resisted my attempts to make it work on a Slackware box. However, it works fine on Debian 3.0 (which is also my preferred Linux distribution.) -- [About a discussion of heavily customised cars.] I thought they were talking about cheap whores - smelly, ugly, brightly coloured, waste of money, and got a cock inside them most of the time. -- Will Hargrave in uknot ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Desired Install in MotorHome
Paul Oster [EMAIL PROTECTED] wrote: I've got a client who [wants VoIP working over a very high-latency link]. So whats everyones opinion, worth exploring further, or am I wasting my time trying? Can you stick an Asterisk box at his end so you can speak IAX over the link? It may not help with the massive delays (which is going to be inherent in any kind of VoIP over the link) but the signalling should be a lot more reliable. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make * don't strip the leading 0
Roger Schreiter [EMAIL PROTECTED] wrote: [...] I have currently the same problem with my E1 card and I wonder, how I can get asterisk to append a leading 0 before forwarding the call, for my IP phones show the correct callee number with leading 0. I ended up just writing a Perl AGI script to canonicalise incoming CLI. -- Hockey has never made much sense to me. In Rugby (my sport of choice, because it's about the only sport where fat, overweight, out of shape guys are actually a sought after commodity), you've got hands, feet, knees, elbows, heads, and teeth. In hockey you've got all that, plus they give you a *STICK*! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Divert to arbitrary number.
Jaun Terblanche [EMAIL PROTECTED] wrote:
I am looking for a way to allow users to dial *21*, followed by a
number and the pound key. Asterisk must then divert all incoming
calls to the user's extension to the number given.
[...]
Dial the Local interface. I have this:
; Macro to place a call to an internal number
; ${ARG1} - Extension (e.g. 300)
[macro-dialexten]
; Look up divert-all number
exten = s,1,DBGet(DIVERT=divertall/${ARG1})
; Dial the diverted-to extension
exten = s,2,Dial(Local/[EMAIL PROTECTED],300)
exten = s,3,Hangup
; if the DBGet fails, we come here - jump to call attempt
exten = s,102,Goto(1001)
; if the Dial fails or is busy, we come here - jump to busy voicemail
exten = s,103,Goto(2001)
On a related note, how do you get a Cisco 7940 to dial numbers with a
hash in them, instead of just using the hash as a dial key. For
example, I have *#21# to check diverts, but the phone will just dial
* as soon as you type the # after it.
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Re: [Asterisk-Users] Divert to arbitrary number.
Jaun Terblanche [EMAIL PROTECTED] wrote: Not sure how to do on the Cisco, but I have found that in certain setups * picks up # as %23. Try putting *%2321%23,1, in the dialplan for *#21#,1, (or whatever your priority is) That's not the problem I have. The dialplan works just fine with a Budgetone. The problem is that the Cisco won't properly dial the number because it has a hash in it. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Divert to arbitrary number.
Philipp von Klitzing [EMAIL PROTECTED] wrote: [...] In case a) you'll have your telco handle the diversion (an ISDN feature) i.e. none of your PBX connected lines will be used. Using the local interface, however, you run the call through Asterisk, consume two lines *and* pay for the 2nd call leg fees. This does assume that your telco supports call deflection, and that they don't charge for the outgoing leg. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Eliminating silence suppression(?) on IAX2 calls
We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI show version Asterisk CVS-05/06/04-18:45:57 built by [EMAIL PROTECTED] on a i686 running Linux Incoming callers are complaining of calls sounding a bit odd. With a bit of experimentation, it seems that there's some form of silence suppression cutting in that clips the sides of words. This doesn't happen on internal calls. Since the problem is limited to external calls, I take it that the problem relates to IAX2 and not some other part of Asterisk? Here's a sanitised version of iax.conf: [general] port=5036 bandwidth=low disallow=all allow=alaw allow=gsm disallow=lpc10 ; Icky sound quality... Mr. Roboto. jitterbuffer=no register = USERNAME:[EMAIL PROTECTED] tos=lowdelay [magrathea] type=friend trunk=yes host=dynamic auth=plaintext username=USERNAME secret=PASSWORD context=inbound-magrathea notransfer=yes Does anybody have any suggestions as to how to clear this odd audio issue, or pointers to what configuration options I should tweak? Ideally, I just want a completely uncompressed 64k channel, as if we were just using ISDN. Thanks in advance. -- What contemptible scoundrel has stolen the cork to my lunch? - W.C. Fields ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM to ISDN or TAPI
Chris Lee [EMAIL PROTECTED] wrote: [...] I am going to have some remote machines which need to have adjustments made to their settings on occasion, the most cost effective and user friendly way I can come up with is a simple IVR system that says press 1 to set limits on flow, press two for flow status report etc. I've got a better idea. Stick the IVR system on a box with a wired telco connection. Not only will this make DTMF more reliable and the audio will be clearer which is useful when reading out lots of statistics, but it will also make it much easier for you to test, manage and update. Unless you like site visits to the middle of nowhere. The backend AGI scripts on this system can then speak a lightweight protocol to the remote system to query it. I'd be inclined to use GPRS as it'd have a very low cost considering the minimal amount of data that would pass, with CSD as a fallback if GPRS fails or the remote handset runs out of credit. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM to ISDN or TAPI
Chris Lee [EMAIL PROTECTED] wrote: [...] I am in the UK and am looking for a device that will allow me to connect two sim cards (read wireless lines) to either the port on the back of my fritz card or any other connection direct to the PC that provides a usable telephony interface. My initial thoughs are to use an X100P plugged into a Premicell, as it's nice and simple, and it would clearly work well with Asterisk. The downside is that it's an analogue connection, of course. -- I want to know how God created this world. I am not interested in this or that phenomenon, in the spectrum of this or that element. I want to know His thoughts; the rest are details. - Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Early Dial
Aaron Martin [EMAIL PROTECTED] wrote:
Has anyone managed to get Early-Dial working with the grandstream
phones?
Yes, but it doesn't play nicely once calls are being gated to the
PSTN.
Early Dial works by attempting a call for each digit that is dialled.
Asterisk will try each such call across the dialplan and will give a
484 response if there's no matching extension but there might be a
matching extension if more digits are dialled.
Unfortunately, this algorithm doesn't work if you have any
variable-length extensions. For example:
# Internal extensions are 300-399
exten = 3XX,1,Macro(dial-exten,${EXTEN})
# Numbers starting 0 are PSTN calls
exten = _0.,1,Macro(dial-pstn,${EXTEN})
If I dial 333, Asterisk will return 484 after the first and second
3. On the third 3, the extension matches, and the call will be
made.
Now suppose I want to call 01234 567890. Asterisk will return 484 for
0. However, when the 1 is dialled, the extension matches, and a
call will immediately be attempted to 01 (and fail), without me
having had opportunity to dial the rest of the number.
Unfortunately, the UK does not have fixed-length numbering. While it
is *mostly* 10 digits (excluding the initial zero), there's some 9 and
7 digit numbers in the mix too.
No problem, I thought. It's just a case of converting the Oftel
codelist.zip into a large exception list of 7 and 9 digit numbers for
the dialplan, and assume 10 digits for everything else. The end result
of this was to discover just how lousy the data was. The number range
containing 0800 800 150 was listed as having ten digit numbers
(remembering that the initial zero doesn't count) as was the somewhat
famous 0800 .
Attempting to kludge round the problem just caused a number of other
issues elsewhere that weren't really Asterisk or the Grandstream's
problem, but made the whole idea of Early Dial impractical for our
setup.
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Re: [Asterisk-Users] Re: Multi process of *
Matteo Brancaleoni [EMAIL PROTECTED] wrote: [...] User Mode Linux is way better for that use, much more efficient. VoIP-only Asterisk also works nicely under vservers (see www.linux-vserver.org), which is even more efficient than UML. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hyperthreading?
Andrew Kohlsmith [EMAIL PROTECTED] wrote: [...] They can't? HT is detected in /proc/cpuinfo (flags) and I see two processors with 2.4.25 SMP kernels... What exactly isn't it using? Linux doesn't realise that scheduling a process onto one virtual CPU reduces the performance available on the other. There can be some quite bizarre scheduling decisions made as a result that can slow things down. On the other hand, for some tasks, it might not cause problems and thus you'll get a boost. As ever, the answer is to benchmark your configuration with and without HT to see. There's no simple answer. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Receptionist manager program.
Kyle Hagan [EMAIL PROTECTED] wrote: [...] We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. Warning, heavy use of touch screens causes gorilla arm and careful ergonomic design of the workspace will be required. Gorilla arm was discovered in the 1980s and is a reason why touch screens aren't ubiquitous, but litigation for workplace injuries is both contemporary and not unusual. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Pascal
usedcanon [EMAIL PROTECTED] wrote: Thanks, suddenly makes sense now. I guessed that is the case however was not sure. Any opinion on what is more/most efficient, using a scripting language like perl or a compile app in C/pascal. Define efficient. A C program would normally be expected to be about ten times faster than a Perl script. But when it's 10ms to execute instead of 100ms, it probably doesn't matter. If your time is not free, it may be more efficient to write a quick script in Perl and buy a faster server than it is to spend ages writing in C. Either way, if you're spending anything bit a trivial amount of CPU time executing AGI scripts (whatever the language), you've probably misdesigned something. So the ultimate answer is that AGI scripts should be written in whatever language you're most comfortable doing them in. -- Vice is its own reward. It is virtue which, if it is to be marketed with consumer appeal, must carry Green Shield stamps. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 11 instead of Star
On Tue, May 25, 2004 at 10:37:25AM -0500, Greg Blakely wrote: [...] + It's just as well that *8# isn't used for call pickup anymore. The # on the end really SHOULD mean end of dialing, and not have any other significance. Unfortunately, BT and GSM service codes give significance to # in the middle of the dialling sequence: *NN# - Enable service with code NN #NN# - Disable service *#NN# - Query status of service Or has this already been discussed to death? Possibly, but some of us are still arguing over the corpse :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 11 instead of Star
On Mon, May 24, 2004 at 07:58:26PM -0700, Paul Crick wrote: [...] *sighs* Yeah, that won't work.. which is a shame.. this goes back to the whole debate of should CLASS service codes be implemented in the dial plan or the channel driver? The most compelling reason to me to have them in the dial plan is that CLASS codes aren't universal. AFAICS, they're mainly limited to the USA and telcos that source cheap switches from the USA and don't bother to customise them. For my Asterisk setup, I'd rather my phone uses the same star codes as BT and GSM than some foreign standard that nobody here knows. From memory and reading the mailing list for a while now, I think Mark's dead against having these features in the dial plan, but I can't remember why. Let me guess, is he American? ;) [...] I think the way it was going to go was a flag which would allow you to disable all channel driver features like this and rely on the dial plan to implement the features. This is very much my preferred solution. If there is still some bizarre obligation to support alien phone standards in the channel drivers, we should have the option of disabling this undesired behaviour. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Link on Web Pages
On Fri, May 21, 2004 at 01:01:10PM -0400, Barry Fawthrop wrote: In Order to place a call Me button on a webpage which would you use ? [A] a href=sip:[EMAIL PROTECTED]Call Me/a [B] a mailto:sip:[EMAIL PROTECTED]Call Me/a I don't know anything much about SIP/VoIP integration within browsers, but I do know that [A] is valid HTML and [B] is not. So [B] will never work, whereas [A] looks credible. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP in the UK
On Mon, May 17, 2004 at 06:12:57PM +0100, Craig Waddington wrote: Voiptalk provide an excellent service and great support. I would hope so, as for many types of calls they're more expensive than BT's basic rates before any discounts! I'll stick to a FXO card and Telediscount/18866, ta. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP in the UK
On Tue, May 18, 2004 at 10:10:49AM +0100, Gavin Hamill wrote: On Tuesday 18 May 2004 09:57, Peter Corlett wrote: [...] I'll stick to a FXO card and Telediscount/18866, ta. Ah, the indirect access ones.. may I point people in the direction of www.1899.com ? 0.5p/min anytime to the UK ... I have enquired if they offer a SIP service - but got no response :) It's a pity their website only seems to work with MSIE :( Hmm, most interesting. The rates seem identical to 18866 apart from that 0.5p/min. I must admit that 1p/min was starting to look expensive - I've been getting that rate since 1999 :) It appears to be yet another front to the Telediscount empire. The complete lack of customer service and the wxx.nl domain doing some backend stuff is a hint. The pricing makes rather a mockery of most VoIP providers! :) This isn't entirely difficult, unfortunately. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP in the UK
On Tue, May 18, 2004 at 10:52:39AM +0100, Chris Stenton wrote: [...] But 1899 is charging 3p setup fee per call where telediscount is not? So depends how long your average call is. There's a 1p connection fee on 18866. So 1899 is cheaper for geographic calls longer than four minutes, and 18866 is cheaper on everything else. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DASS2 support
My employer wants to use Asterisk, but the E1 circuit providing the current phone system is DASS2 rather than ISDN30. Converting the E1 to ISDN30 is not a practical proposition at this time as it'd stop the legacy phone system from working. Is there any sort of hardware support for DASS2? I speculate that the E100P should be able to deal with the electrical side of it, but I'm unsure of driver support. Has anybody got Asterisk to work with DASS2 circuits? Thanks in advance. -- There are three reasons for becoming a writer: the first is that you need the money; the second that you have something to say that you think the world should know; the third is that you can't think what to do with the long winter evenings. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
