RE: [Asterisk-Users] Zaptel: Hz != 1000 causing ztdummy compilationerror?
A jiffy is a kernel timer, this affects many thing in the kernel. Linux for as long as I know uses 1000hz. I am really surprised this failed on fc4. Ztdummy uses this as a base for timing, particularly with meetme and tdmoe. If its not high enough quality may be degraded. As for what you tried, you tried to adjust the realtime clock, which is slightly different. What kernel version are you using? -Original Message- From: Patrick[EMAIL PROTECTED] Sent: 11/4/05 2:34:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Zaptel: Hz != 1000 causing ztdummy compilationerror? Hi all, When I compile zaptel from today's cvs HEAD on an updated FC4 box it fails with the following message: CC [M] /home/patrick/redhat/BUILD/zaptel/ztdummy.o /home/patrick/redhat/BUILD/zaptel/ztdummy.c:103:2: error: #error ztdummy requires 1000 hz jiffies If I comment out the code causing that error the compilation goes fine but I guess it's there for a reason :) After some googling I tried the following but that did not solve the issue: echo 1000 /proc/sys/dev/rtc/max-user-freq compile still fails echo 1024 /proc/sys/dev/rtc/max-user-freq compile also fails Anyone have a pointer how I solve this error? Thanks and regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [Message truncated. Tap Edit-Mark for Download to get remaining portion.] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX test service
http://www.0xdecafbad.com/Free-VoIP-Providers.htmlhas a list of some free providers -Original Message- From: Gabor Horvath[EMAIL PROTECTED] Sent: 11/3/05 3:13:07 AM To: Asterisk-Users listasterisk-users@lists.digium.com Subject: [Asterisk-Users] IAX test service Dear Asterisk users, can you suggest me a free service where I can test my IAX trunks? Thank you. Gabor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users [Message truncated. Tap Edit-Mark for Download to get remaining portion.] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
On Thu, 2005-10-13 at 14:03 +0800, Craig Guy wrote: I have downloaded iaxmodem and gone through the readme but not yet installed it. I currently use rxfax to receive in the vicinity of 1200 faxes per day and 5000 or more pages (faxes vary from single page to 30 pages) per E1, with a peak load of about 12 concurrent inbound faxes to rxfax. Best I can tell my failure rate is about 0.8%. I have been testing using Hylafax for faxout with an 8 port analog fax modem card and a couple PAP2NA's and this works well, but I am very much looking forward to checking out iaxmodem. Especially if using Hylafax will give me ECM. Craig You may have already planned this, but I would be interested in hearing how it works for you. Granted that will take some time for you to even know how well it works ... As a side note I am looking at iaxmodem now (although I am easily distracted) with the hopes of using some of the modem codecs spandsp supports to at least get tdd support working for asterisk, and the end hope of more modem protocols. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wanting to Make a PocketPC have a secureConnection to asterisk server
On Thu, 2005-10-13 at 08:16 +0100, Steve Daniels wrote: VPN? IAX and an SSH Tunnel? Does anyone know of a good solution to create a secure (encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk server? Pocket pc supports VPNs natively. No additional software required, assuming you have something on the server that can talk to it. What that is specifically I dont know but perhaps google can tell you what vpn solutions work with the pocket pc. Its not going to be totally secure, with crypto the questions to answer is 'secure from whom and for how long'. Odds are it will be secure enough for the types of data you would have and the types of people that would likely be in a position to eavesdrop. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Application: Broadcast
On Thu, 2005-10-13 at 08:20 +0100, Steve Daniels wrote: What excatly does it do? What messages does it send out? And what software needs to be configured to listen for these messages? Answer these questions and maybe more people will download the source :-) As was explained to me via private email you would do something like: exten = s,1,answer exten = s,2,broadcast(some arbitrary message here) exten = s,3,blah any of the configured systems would get the message, so if you wanted to broadcast caller id or anything else from within a dialplan you could. Any arbitrary message can be embedded in any dialplan wherever you want. As for the listening that wasnt asked by me nor answered (hard to answer a question that is never asked :) so I cant say. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Email to FAX
On Thu, 2005-10-13 at 09:18 +0200, Bohuslav Coufal wrote: Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Effectively T.37 does that, however what you prolly wanna look at is hylafax to process the emails (perhaps by procmail). From within asterisk how the call gets placed doesnt matter a whole lot, and you do have options but basically what you need is a modem (physical of soft like iaxmodem) and a phone line to transmit (analog or digital bri/e1/t1/j1/etc). If you want the call to go through asterisk you may need slightly more, some way to inject the call into asterisk (FXS port, iaxmodem, whatever, all depending on how you configure the device to receive the faxes by email). You can also outsource, some random sites are listed at http://www.voip-info.org/wiki/view/Asterisk+Email+to+Faxl -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
On Thu, 2005-10-13 at 15:55 +0800, Craig Guy wrote: I'm trying to figure out what an appropriate deployment model might be. Whether to have iaxmodem installed on the hylafax server with a switched ethernet connection for iax2 to the * server with the PRI, or to have the iaxmodem on the PRI * server and channel the tty comms across the network. I suspect that the latter might work ok over a WAN so I could have a central hylafax server with distributed * servers running iaxmodem at the far end of wan links (up to 100ms latency). The only issue is that I want to retain rxfax on the PRI * servers for incoming faxes. Based on the docs in iaxmodem its better to have iaxmodem on your asterisk server and hylafax (if needed) on a remote server. The lag issues between iaxmodem and asterisk are more critical than hylafax and iaxmodem. Lee, if I install iaxmodem on a * server for outbound faxing from hylafax, can I still use rxfax on the same server to receive faxes? IAXModem works like an iax client, if you redirect calls to that extension they goto iaxmodem if you dont they are handled elsewhere. Treat that as just another extension for all intents and purposes. Problems however may arise if asterisk is told to redirect all calls with a fax tone to rxfax, so you have to deal with that in your dialplan. You would have to either get clever with the extension or do did based routing ... exten = fax,1,gotoif(something?2:3) exten = fax,2,rxfax(somefile) exten = fax,3,Dial(IAX2/iaxmodemExt,60,R) Although this isnt an issue if you do did based routing and the given did is one or the other for that context but not both. Hope this helps (and I hope I am right, but I have been reading a lot and think I am, I am sure lee will point out anything I got wrong) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD/queues question
Just remember to set your phone in the group with the highest possible priority :) On Thu, 2005-10-13 at 09:36 +0100, Pedro Nunes wrote: Thanks, That will fix my problem... And agent skills, is that possible too?? Thanks again Pedro Nunes -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Email to FAX
On Thu, 2005-10-13 at 10:45 +0200, Coufal Bohuslav wrote: Hi, when I try to send fax by example in README I got nothing. On asterisk console i saw this: -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) Channel Zap/4-1 was answered. Launching txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) on Zap/4-1 -- Hungup 'Zap/4-1' http://soft-switch.org/installing-spandsp.html When sending a fax it is more likely you will be calling out to the remote FAX machine. In this case, make your Asterisk call the far FAX machine, and when it answers do: exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller) The addition of |caller will make txfax act as a calling machine, rather than an answering machine. This seems ti imply that txfax() doesnt actually dial anything, you have to do that elsewhere, I suggest you use the outgoing spool directory and place (mv not cp) a file in there. Channel: Zap/1/5551212 Maxretries: 0 Waittime: 20 Application: txfax Data: /tmp/fax.tiff|caller This will cause asterisk to call on Zap/1 and dial the number 5551212, when that answers it will call txfax and pass it the path to the fax file and caller (so it acts like a caller not a server/answering endpoint). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
On Wed, 2005-10-12 at 16:48 -0500, Tim Litwiller wrote: See IAXModem above for the soft DSP. There is very little info on the sf.net page regarding its capabilities ... Does it only do fax or does it do other data communications? What fax protocols are supported? Does the destination path from asterisk-whatever need to be iax or will asterisk properly translate to a different medium (eg presumably iaxmodem does iax to asterisk, then from asterisk does it matter if you use sip, zap, h.323, whatever ?) I cant see where it would matter once it hits asterisk, but stranger things have happened ... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk logo
On Wed, 2005-10-12 at 08:59 +, Andrew Nowrot wrote: Hi, I was wondering if I could use Asterisk logo in my PBX system which I want to introduce in my local market. Does anyone know if I must fill some legal issues which let me use this logo. Best regards digium is the owner of that, they are revamping (may have completed that) the document which describes when and where you can use it. You should contact digium directly to see if they are ready with their new terms for use (the logo is trademarked). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
On Wed, 2005-10-12 at 18:18 -0400, Tom Rymes wrote: On Oct 12, 2005, at 11:26 AM, Lee Howard wrote: If your PRI comes in to a TE405P or somesuch then you can pass fax DIDs out through another port on the TE405P and out to a T1 faxmodem (such as a Patton 2977) or a T1 channel bank and then to analog modems. Good call, Lee. Unfortunately, we only have a single port Sangoma card in our asterisk server. In order to do what you suggest, I would have to buy a dual port card and a channel bank or T1 modem. Thats more $$$ than is warranted by our fax traffic. Also, given reports of problems related to frame-slippage and other weirdness encountered when sending data/fax through Asterisk, I'm reluctant to invest that money. Have you tried this setup yourself? Cant iaxmodem work by having asterisk bridge the pri channel as needed (did based routing perhaps) and then have hylafax use iaxmodem as the modem it uses. That should result in no additional hardware, which means testing can happen with little cost to see if it works for you. As I understand it iaxmodem just acts like a modem, and doesnt actually do the processing that hylafax does, so the two would work together instead of one or the other. I may be wrong on this, but that is the way it looks to me so far. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
On Wed, 2005-10-12 at 18:45 -0400, Tom Rymes wrote: This is true, but: 1.) Lee has stated that IAXModem is still Developer-grade code. 2.) I don't have a spare PRI for testing, and our phone system is far too mission critical for me to go mucking about with it and trying this out (especially given #1, above). 3.) It will not be easy for me to test out this setup without simply switching our production HylaFAX server to use IAXModem, which I am again reluctant to do, seeing as it is our production server and we depend on it. Testing fax service setups is notoriously difficult due to the huge number of different fax machines, etc that are out there. redirect one did to iaxmodem for now, test it out, you shouldnt have to reconfigure everything to get this to work. iaxmodem connects via iax so it acts like any other client in that regard, so you should just set up an acct and redir an unused did to it, assuming you have an unused did of course. Would that not solve in the short term all of those issues or am I missing something? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] parameters documentation
On Wed, 2005-10-12 at 09:41 -0400, Time Bandit wrote: Is there a place where all the parameters are documented ? In example (my example!) I would like to know the meaning of a lot of parameter that can be used in sip.conf, http://www.voip-info.org/wiki-Asterisk+config+sip.conf How did I found this ? http://www.google.ca/search?hl=enq=site%3Avoip-info.org+sip.confbtnG=Google+Searchmeta= Remember : google is your friend to elaborate slightly ... if you type into google site:voip-info.org asterisk type item where type is cmd or config and item is either the config file name or the command you should be able to get there. Alternatively you can just straight there by entering the url: http://www.voip-info.org/wiki-Asterisk+TYPE+ITEM Google is handy if you dont know the name of the command in question because you can just omit item and it will show all the commands available :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
On Wed, 2005-10-12 at 19:19 -0400, Tom Rymes wrote: Would that not solve in the short term all of those issues or am I missing something? Well, I can redirect a DID to it, but I have no fax traffic going to that DID, and I am still reluctant to install developer-grade code on my production asterisk server. The idea of redirecting an unused did is so that you can develop your test cases then see if the code works how you expected it. I would hope that you wouldnt have any real traffic aside from your test cases :) As for development code, I can understand that, and is actually a good practice to only use stable stuff... However remember that it is open source and often it stays in development much longer than most companies selling a product keep code in the dev stages. This is because its not being sold so there isnt market pressure to make it 'stable'. Far too often commercial products (not all but enough) release 'stable' products that are far from it, infact they act more like they are in the final beta stages ... Perhaps Lee can comment on exactly how 'development grade' it really is, perhaps even cite some test cases where people have used it on larger scale operations (ie larger than a home users 1-2 times a month or less). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soekris and Asterisk
On Wed, 2005-10-12 at 17:46 -0700, Paul Mahler wrote: You need about 30MHz per channel. That means the Soekris can only handle part of a T1, it will never handle a quad span. Paul How was that determined? I have a problem with a plain number like that, which may have been taken into account, why I am asking... Different cpus operate differently, taking more or less time to complete certain functions. Instruction optimization can go a long way if those instructions are used (not terribly likely if its just pushing bits but there are some for just that). Additionally there is no codec processing (presumably) with TDMoE, does the 30MHz take into account any codec processing or is it literally 30MHz (on what cpu class?!) for just pushing bits? There are other factors, but you did say 'about' so they are optional to this conversation, ie other IRQs on the box, potential for device polling, etc. A tuned system for that specific task (pushing bits between a TDM card and ethernet via TDMoE) may be able to operate at a lower clock speed per channel, but that isnt as important for the initial questions. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Modifying cmd VoicemailMain
On Thu, 2005-10-13 at 10:08 +0900, Kuniyoshi Murata wrote: Andy Kuo writes: Hi, Maybe you can record the sound file vm-five.gsm as five hour in Japanese, instead of just five. AK I don't think you can do that. Because that vm-five.gsm can be used as message number also (e.g. message FIVE) For the other changes I am starting to think that it will require either modifying the voicemail app or doing voicemail as an agi or dialplan setup. All 3 have some drawbacks, but would give you the ability to tweak everything exactly how you want it. As either an agi or dialplan setting you could use most of the voicemail app functionality if that is suitable (I dont know where the prompts are exactly that the original poster refered to). It may boil down to writing a complete voicemail system as an agi or modifying the voicemail app to get exactly what is wanted. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I use different languages (Chinese, Cantoneese)?
On Thu, 2005-10-13 at 12:17 +0800, Ronald Wiplinger wrote: I want to give the users the announcements as they subscribed to. The announcements should be in English, Chinese, Cantonese, according to their phone number. How can I do that? I can hardly make for each number a different context!!! http://www.voip-info.org/wiki-Asterisk+cmd+SetLanguage either use an agi to fetch the settings, or dbget() -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 07:53 +0100, Chris Bagnall wrote: For the UK, your most accurate source of info is probably the first few pages of the BT Phone Book (delivered free to all UK homes/businesses). There's quite a comprehensive list of what each number range is for, how it breaks down, etc. Ok, then let me ask the obvious, can anyone with a phone book publish that info? Although there its fairly easy restrict everything that doesnt start with a 1 2 and you are safe, start opening that up and you may or may not be depending on provider (some that include uk landlines do not incl freephone for example). I did find a list somewhere of the useful special services (1xxx.) such as BT customer services, line tests, etc. that I can copy from one of our clients' dialplans if it'd be any help? dont most of those start with 14? like to enable/disable caller id and such. But regardless I am more interested in what is valid landlines in various countries, not just the UK but all countries. When trying to create a dialplan for voipbuster to avoid charges I noticed that several of the countries didnt appear to have identifable mobile numbers (at least according to wikipedia) and others werent listed at all. This means that there are several countries I cant tell. But voipbuster aside this information is generally a good thing to have for anyone who wants to either use a voip service provider and not be charged the sometimes insane termination rates to mobiles, or worse call a premium service number for $2511/minute (as was the case with a +1 809 number once). Then for the voip provider side, there are rates sheets but sometimes they are incomplete on what is what, and carriers have pass through billing. Take UK premium most people list those as 44 9xx or 44 90x, but both of those arent entirely correct. 44 945 is a (now deprecated) pager prefix (or was that 941?). 4490x and 91x are premium service numbers, block only 90x and you miss the 91x which can be as much as 1.50 GBP/min (and despite the hype icstis does have exemptions on certification for those that get a premium number even at the 1.50 GBP/min rate). The cost per call to some UK premium can be quite high when its a 'call for payment of a product' type service. Then take into account other countries with looser telecom laws (liechtenstein, afganistan are two that come immediatly to mind). People have been setting up phone companies there then changing the published rate for termination to their phone company and with pass through billing the voip provider gets hosed unless they too have pass through to their end user (I bet that will become more popular in the near future - although by the time it happens the credit/debit card can be canceled etc). This is basically what happened to nuphone to the tune of $450k in one month. All of these factors need to be addressed in a good dialplan, although its really hard to keep one as dynamic as it would need to be (ie someone sets up a new phone company, you need to know it exists, maybe mark it as 'new' for a while to see if any suspicous traffic comes in - if you can even tell if its new). With other countries where the phone numbering system is fairly static and fairly regulated the lists shouldnt be that hard to create. This would be a nice wiki page on voip-info.org if people contribute the numbers from their localities to at least get it started, and over time it would grow to something quite usable. I used to have a itu.org or something page that had number formatting for each country although I dont think that it went into what is and what wasnt mobile, special services, and landlines. will keep trying to find that again. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 09:34 +0100, Steve Kennedy wrote: 448xx national rate, local rate, freephone, some mobile, blah 44800 is freephone and 808 and um what is the third? I wanna say 500 but I am not sure that is right. there is an on-going discussion whether 4487 numbers (or at least some) should go into premium rate. Well legally they arent yet. But there are some 87 numbers that are for mercury, some are vodaphone, cellnet, BT, um um um. Then you have 871 (10ppm national rate) 870 (NCFA) 87 has more than what is most popular (870/871) and that makes it a problem depending on the rates to the specific 'subexchanges'. Note about 871, the 10ppm is the BT rate, one voip provider is about half that while some others traditional carriers are 10-40% more. Although I dont see something that states what a valid number actually is. So idealy to avoid getting charged a higher rate I would want to limit all calls to the UK to region codes starting with a 1 or 2 (although from what I have seen most of the 2x is 20 for london). There are other areas too covered by 2. Yeah thus the use of the word 'most' :) Also you have to know who's terminating it. You can make an assumption re BT termination, but directly connected businesses may use another telco with different termination rates etc. A lot of UK telco's have difficulty pricing UK termination because of this (or risk taking a hit if they get it wrong). Yeah, although some companies (CW for example) have built out their own network and can terminate cheaper than BT so ... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 09:36 +0100, Steve Kennedy wrote: On Tue, Oct 11, 2005 at 07:53:09AM +0100, Chris Bagnall wrote: For the UK, your most accurate source of info is probably the first few pages of the BT Phone Book (delivered free to all UK homes/businesses). There's quite a comprehensive list of what each number range is for, how it breaks down, etc. I did find a list somewhere of the useful special services (1xxx.) such as BT customer services, line tests, etc. that I can copy from one of our clients' dialplans if it'd be any help? Actually the best indication is Ofcom's site (the regulator), they publish number blocks and who owns them (doesn't take porting into account, but it's a start). www.ofcom.org.uk Yeah but that still doesnt answer the fundamental question. While they do have who owns stuff they do it based on prefix (ie 44 871 59 is pipemedia) but it doesnt tell you how many digits are past that (5 more with that one anyway). I dont know that all UK phone numbers are 11 digits (infact I am led to believe they arent). And outside the UK you may want to know what is premium, mobile, etc. What would incur higher charges for the call itself. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 10:50 +0100, Are wrote: in AstBill the MySQL table 'astcountrycode' contain 601 records of countries and US states including the countrycode and US State Codes. I will look at that, although personally I dont like drupal, and dont need a billing system per se, I had planned onl ooking at it for other reasons. Where do you get the data from? Obviously I can just dump it but it would be nice for me to know where you got it from. I did notice this: http://www.voip-info.org/wiki/view/Numbering+plans which links to http://www.itu.int/ITU-T/inr/nnp/index.html (listing of basically every countries numbering plan administrator) and also links to http://www.wtng.info/ (not what I had hoped but for now it seems to be the best thing going ...) Maybe I will make some parser that will deal with the data from astbill although I want to make sure of its origin (no offense, but there is a ton of info on the net as a whole it cant all be true :) and make sure that its current (from what I have seen on www.itu.int a few countries are renumbering lately. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 11:40 +0100, Chris Bagnall wrote: AstBill the Web-based open source Billing and Management software for Asterisk includes the information you are requesting. big snip Apologies for the slight threadjack, but as someone fairly new to the list, what *is* the policy on list advertising? There are quite a few posts I've seen in the few weeks I've subscribed that are pretty close to the blurry line. Back to the original topic - what's the purpose for which you need this information? Building dialplans or billing? personally dialplans, for some associates of mine they want billing abilities, more specifically they have a desire to know better what is what in a foreign country. While they know their own numbering plan, and a couple of other countries there are a BUNCH of countries out there with totally different numbering schemes. Others on the list may want it for either, I have a feeling this request goes beyond my personal desire. What I did was get astbill, I am trying to see where all the bits of info is that I need (starting to look like two tables I will have to go through to get everything I want). I may make a page that will create extensions.conf cut-and-paste stuff tomorrow (I am still up from yesterday so odds are not tonight - the sun is almost up I am about to turn into a pumpkin :) Assuming astbill's data is accurate, and my very very brief view of it based on the limited countries I do know of it appears to be fairly complete, then people should be able to goto my webpage and either download the data and run it themselves or possibly use a webapp on my webpage to create an asterisk dialplan for specified countries based on a very simple template. I may make it simplier with instructions on sed, it all depends on how I feel. If you save the file and run sed it would be far easier for me :) As you've discovered, the UK is a complete mix of different number ranges, often numbers within those being billed at different rates. It used to be It started with the UK as an *example* and everyone seems to have latched onto that. I wanted to know more than the UK, I wanted every country. astbill seems to have that data, I seem to have located all the little bits that I need from that data, so this is progressing along, from my perspective anyway. 449xx premium services Most of our clients generally ask me to block everything staring 09xx by default. The few pager numbers that are still in this range should (I'd be interested to hear if they're not) have been assigned new ranges in the 07xxx block by now. I have gotten through to 1 pager in the 945 or 941 I forget now in the last week or so. You think country code 44 is a mess, think about country code 1, it spans many countries ... some in +1 have had $2511/minute rates. Yes twenty five hundred eleven united states dollars per minute! Country code 1 is really a region code (north america) and becuase of the different countries there are different rates, interconnection fees, laws governing the numbers, etc. So it could be worse :P -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 14:15 +0300, oner asterisk wrote: most compherensive list that I saw is http://www.numberingplans.com/?page=diallingsub=areacodes The only problem with that is in the first one I tried (Ireland) it didnt give me the specific info I wanted :/ I tried the US and it is a really short list and doesnt include numbers like 700, 900, tollfree, etc. In ireland (+353) mobiles are 083 085 086 087 088 with 088 largely deprecated (eircell analogue). New subscriptions have to be from the assigned pool, so there havent been any new 088s issued for a while, maybe someone ported to digital service? I dunno, its mostly deprecated anyway :P I would like to know if a call is a landline, premium, mobile or other type of special service. It appears that astbill has this data, and I am working off that for now, I would just feel more comfortable if I knew where they got it from (not enough time has passed from the first time I asked for that, so I dont fault them for not saying anything just yet). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi Phones
On Tue, 2005-10-11 at 13:36 +0200, Patrick wrote: On Sat, 2005-10-08 at 18:01 -0400, Cory Andrews wrote: The F3000 is not anticipated to be available for distribution until late December/January, FYI. I came across this one. Haven't seen one in real life though. http://www.gemtek.com.tw/pro_whsg103g.htm Regards, Patrick Just make sure you have plenty of bandwidth available, keep in mind wifi is half duplex so it gets full a little faster than you may think. Then stuff like jitter buffers and all make a big difference in performance. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid validation and expression parser problems on Solaris 10
On Tue, 2005-10-11 at 14:12 +0200, Joseph Rothstein wrote:
I am setting up a voicemail (VM) system based on Asterisk. From what
I've heard Vonage uses Asterisk as their VM platform as well. I am
running 1.2beta with a MYSQL backend for extensions and VM user info.
All the sound files and vm messages are being stored through an NFS
mount externally. The reason for this is that there will be several
asterisk VM frontends, all accessing the same config and vm user info
as well as sounds files.
By sound files do you mean the static 'enter your password' type files?
You may do a lot better having those on the individual boxes to reduce
the load on the network. If its static keep it on the box, as a general
rule.
priority='5',app='gotoif',appdata='$[${sanity}]?10:20';
goto 10 if true 20 if false
[...]
-- Executing set(SIP/10.10.13.110-00123d48, sanity=0)
0 means false
[...]
-- Goto (default,03413306999,20)
goes to 20 as you told it to
Sanity is just a variable to keep track of whether or not cid and vmid
are equal. IN this case they are, so the statement
no they arent.
priority='1',app='Set',appdata='vmid=03413306999';
priority='2',app='Set',appdata='cid=03413306990';
6990 vs 6999
If anyone has come across this problem, and has a solution I would
very much appreciate any input.
Many times I have been victim of typos. My solution has always been to
fix the typo then proceed. Hopefully that solution works for you too :)
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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Re: [Asterisk-Users] callerid validation and expression
On Tue, 2005-10-11 at 17:06 +0200, Joseph Rothstein wrote: Thanks for the reply Bret. I have tested this parsing issue ever way possible, equal variables unequal variables, arithmetic, ie $[1 + 1], etc., etc., and my conclusion is that Solaris does not parse the $[expr1 operator expr2] function properly. It always produces a value of 0. I installed 1.2beta on a SUSE box and it works flawlessly. Thanks, Joe Ok, I only saw what you posted, where the numbers were not equal and it did what was expected in that scenario. I cant say for solaris with 1.2-beta otherwise. I find it very odd that it works on one operating system but not on another given that that part of the code should be platform independant. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 16:47 +0100, Steve Kennedy wrote: And outside the UK you may want to know what is premium, mobile, etc. What would incur higher charges for the call itself. You can find that out from the Ofcom number plans They have lists for outside the UK? I didnt think they did that ;) At any rate I ripped the database tables from astbill.com and am using that as a base (found a few missing ones too, which are now submitted to them :) I am almost done with my super wonder magical include for adding countries easier than ever to extensions.conf (bet it gets out of date by mid next week). I will try to toss this on my page today, under the voip section, along with directions on how to use it and why I did it the way I did. In essence each country is a context, there are super contexts for country code 1, a US48, US50, Canada, caribbean, and pacific and of course all of country code 1. Each state in the US, and region in canada as well as all the different other countries tossed into 'country code' 1 are seperate if you want to pick and choose. I am trying to make this complete, however expect errors, omissions and general breakage, if it works for you hey great. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 11:16 -0500, Kevin Scott wrote: I'm curious, for a $2511/min call, which +1 number was this? +1900? Kevin 809. It was set up as a premium number in that country. While that was an extreme case it did aparently happen back in the 90s sometime. And because the country in question (I forget exactly which one) didnt require it there was no recording at the begining of the call saying that it was a premium number, nor anything saying what the rate was. Because of the way things ended up the telcos told the company there they had to collect themselves (they didnt) and didnt charge their customers. It was in a few newspapers (I read about it in the NJ Star Ledger at the time, but that may have been an AP story I dont know now, just that was the only paper I read at that time and I know roughly where I lived when I read it, and do recall it was back when I acutally read paper based newspapers). Most of the scammers are smarter than to try something to insane, get a few dollars here and there, so it is a legitimate problem. I have only read one story with that amount, but I would imagine that it goes on far more, just with a $1-2 per minute charge instead ... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country code list
On Tue, 2005-10-11 at 19:54 +0100, Bob Goddard wrote: On Tuesday 11 Oct 2005 12:19, trixter http://www.0xdecafbad.com wrote: [...] It started with the UK as an *example* and everyone seems to have latched onto that. I wanted to know more than the UK, I wanted every country. astbill seems to have that data, I seem to have located all the little bits that I need from that data, so this is progressing along, from my perspective anyway. [...] Try http://global.mci.com/uk/customer/. Look at the box in the bottom right hand corner. It may give you a bit more info. Thank you this is what I wanted all along. Yes they do have more of a breakdown, at least with mexico. I am unsure if I will use anything from astbill, not becuase the data is inherently bad but because I may change the way I do this ... I am thinking parse a CSV and create the dialplan out of that, that way its easier to regenerate in the future. Comments, spacing and everything my dialplan from astbill.com was 1700 lines or so, the MCI file is about 4300 lines, so there was definately stuff missing (whether that is astcill or my that was missing from what was there I dont know). I should have something all new and better soon for people that wanted to create better dialplans. And to think this whole thing started out personally because I wanted the ability to create a good dialplan for voipbuster.com and some friends wanted me to think over billing solutions (not for a real product, a friend is mentoring a high school student in independant study on voip). Thanks a lot for this :D -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
On Tue, 2005-10-11 at 17:01 -0400, Tom Rymes wrote: Frankly, I would recommend that you forget about trying to fax with Asterisk. Buy a good Multitech analog modem and install HylaFAX. Use the right tool for the job!!! Asterisk may be able to fax better in the somewhat near future. One of the things holding up T.38 support is the inability for asterisk to switch codecs on the fly. I am not saying that is the only thing, just one of the things. Well 1.2 is supposed to have better support in that regard, which means that work on T.38 can happen in a better way in the future. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Large country based dialplan
I spent the last day or so gathering every country plan and listing prefixes as mobile, premium, etc. If anyone wants this I have made it available at http://www.0xdecafbad.com/Global-Numbering-Plan.html Each country has its own context, making it easy to include what you want where. Obviously this is not good for enterprise solutions, so I have also provided a csv file for easy MySQL insertion of all the same info. For country code 1 I have broken it down into each country, for the US and Canada for each region, and then made contexts that include those so it should be fairly easy to include into what you need. Cutting and pasting would be a far better solution than including this whole file as it has 5000 entries. But for those that use something like voipbuster that gives free landline calls to a few countries this may be helpful to know what you should be able to call and what you should. Where I got the data from and all is also on that page if anyone wanted to make their own lists. I would appreciate any updates or corrections that anyone happens to notice. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk certification - thread hijack
On Mon, 2005-10-10 at 14:16 +0800, Craig Guy wrote: The practical part of the exam showed a distinct USA bias - It was in terms of T1's and analog zap extensions. I am from Australia, and the exam was in That is ok, most of this list seems to be the same way regarding the US/North American Bias :P I do agree that there should be regionalized tests, for pstn parts, which leaves (aparently) the bulk of the test standardized for the rest of the world, given that the VoIP parts, cli, etc are all going to be the same from that point on. I see certification a good thing for digium, perhaps more than for those that get certified. If a bunch of people are certified then it shows 'market acceptance' further if digium takes care of those that get certified then they are far more likely to recommend digium products, whether that is asterisk business edition (presumably because the gpl version isnt suitable for that customer) or digium hardware. If digium turns its back on people that get certified then they may decide to go with a different provider for hardware and such. The testing I am sure is fairly new, and recommendations like that could go a long way, of course you have to end up with competent people to actually write the test and ensure that its accurate and meaningful. This may be the larger part of the problem, but certainly not one that is that hard to overcome. I think some certified logo would be a nice thing, to help promote both asterisk as well as certify that people are indeed certified, although it would require some backing by digium to make that hpapen (unless a testing system is done 3rd party). That way people can verify that the individual really is certified. I think a 'find a certified asterisk expert' tie in would be a good thing for potential customers or whatever, they goto digiums site, see a listing of all the certified people, and have a url and/or email contact info so they can pick someone, however from digiums perspcetive that would create a potential liability issue in some parts of the world where sueing if anything doesnt work 100% the way they hoped, saying digium 'recommended' the vendor who caused them problems. So a big legal disclaimer is required which can put a bad impression to those reading that page. Its a quagmire. You brought up some good points with all of that, points that digium can potentially address in the future, and I recommend anyone else that feels the way you do to email digium directly, offlist, with their concerns regarding the certification process, mnaybe that would cause the fastest change. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astricon Podcasts?
On Mon, 2005-10-10 at 23:10 -0400, Dean Collins wrote: Yep, I'm stunned that as a technical social network we're not leveraging the technology through webcasts/online presentation, dial in conference calls for the sessions etc. But they charge admission for astricon, who would pay for the dial in conference? :P -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astricon Podcasts?
On Tue, 2005-10-11 at 00:05 -0400, Dean Collins wrote: The question is would people choose not to go if it was necessarily available as a broadcast. You're thinking old school. Dean I am thinking the convention was set up for money, I cant believe that the rate generates no profit. Not that profit is a bad thing, but anyone doing something for profit isnt going to stab themselves in the back to prevent that profit. There is added value to go in person, you get to have side conversations, do networking, get to see the slide shows (which can be done via a webpage) etc. But there is something better about being there in person. So I believe people would go, but maybe not as many, and there is a cost to providing it voip style, bandwidth, servers, etc. Of course if you really wanted to be clever you would have several nodes that people call into, which are all connected to the main server that is at the conroom hooked up to the microphone, etc. That way all the traffic to the main server is stable and relatively low, and the leaf nodes (other asterisk boxes) have the brunt of all the traffic. This way the hub server would not be flooded off ruining everything for everyone when 134091309451093 people try to connect to it. Adding the record functionality and muting participants would also mean that the hub server would be able to make audio files available after the lecture is over. The main server could run a shoutcast stream to be fed to mp3player() or something on the leafs (idealy you would want a proxy on the leafs so each leaf causes 1 and only 1 stream off the main hub. Could be a good marketing tool. Tout the final number of clients connected in such a distributed environment listening live. Show the power to skeptics. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] country code list
I was wondering if anyone has put together a comprehensive list (that is reasonably maintained) that lists country codes, landline numbers, mobile numbers, etc. The particular requirement is for a dialplan to know what is going to be charged to whom. For example, mobile and landline rates are the same in the US the US has a unified numbering plan of 1NXXNXX, while the UK has: 441xxx geographic based landline 442xxx geographic based landline 443xx reserved 444xx reserved 445xx corp and voip 446xx reserved 447xx pagers, personal etc 448xx national rate, local rate, freephone, some mobile, blah 449xx premium services Although I dont see something that states what a valid number actually is. So idealy to avoid getting charged a higher rate I would want to limit all calls to the UK to region codes starting with a 1 or 2 (although from what I have seen most of the 2x is 20 for london). I have found http://en.wikipedia.org/wiki/Area_code but I dont know if its accurate, and would prefer a more authoritative source for the information. I also dont know how out of date that is, some countries have easier telecom laws that allow people to set up phone companies then change the rate form what was published for termination. I certainly dont wanna get stuck with a pass through billing situation on those. If anyone has a very well maintained list somewhere (voip-info didnt seem to have anything, at least not the keywords I have tried with google, admittedly I havent spent much time yet) ... This has to be an issue that voip resellers have to deal with so there must be lists somewhere, although odds are those are rate sheets from the actual provider doing the termination since very few countries own the wire all the way to all those foreign countries, although more are starting to. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Podcasts?
On Tue, 2005-10-11 at 12:57 +0800, Dinesh Nair wrote: On 10/11/05 12:34 trixter http://www.0xdecafbad.com said the following: Adding the record functionality and muting participants would also mean that the hub server would be able to make audio files available after i'd think that muting would be a prerequisite, even if recording was not done. it'd be audio bedlam otherwise, and the speakers would be drowned out. connected in such a distributed environment listening live. Show the power to skeptics. we had such ideas to use asterisk to broadcast our recent HackInTheBox Security Conference (conference.hackinthebox.org), but bandwidth prices at the venue were too high to make this viable, given that it's not revenue generating. corporate sponsors :P Aside from that there are alternatives if you are just doing a one way stream. With the proper gear wifi can carry a signal a considerable distance, providing you can get the elevation on one end or the other, or both (easiest since total height is divided between the two sites). Most venues dont like people rigging up a c band dish in the swimming pool area though :P Then feed that to some site that is more remote than the venue, perhaps a home or office of a local person, who gets the feeds to a bigger badder server. If doing one way latency and all that isnt that big of an issue and you dont need that much bandwidth. If you were to only shoutcast streams at telephone quality you could easily do that over dialup. There are $10/mo tollfree dialup providers in the US that could be used. 1 stream which feeds a bigger server that handles all the clients. Or depending on need, one stream off dialup to a server that feeds 5+ leaf nodes where the end users connect to. If doing it asterisk style you can use mp3player() within asterisk to connect to the aggregator system (ie what dialup feeds) or even the leafs if you are big enough, yes there will be some delay, but it would still work, however complex this has gotten. http://lbtech.com/dialup/ (I am not affiliated with them just know they advertise what I claimed earlier). Monthly cost - $9.95 (NO additional fees or taxes, no matter how much you use the connection) All off a US tollfree. Could work to get the feeds out of the building to a server somewhere to distribute that as needed in whatever formats are required. And if its a lecture hall, a direct feed from the microphone into a system that does the streaming, you only need mono and low quality bitrate for it to be quite acceptable. In theory, you could do several lecture halls at the same time off one system with one inet connection, sound gear would be the hardest thing for a laptop. Maybe usb/BT audio devices given limited port spaces on laptops. Maybe multiple laptops doing wifi or whatever to each other to share that connection. Even if it costs a small setup fee to get the outside line from a conference hall (they will normally charge at least per outbound call, if not a fee to have a line activated in the hall itself) the total cost should be well under $20 to provide this, plus whatever it takes to distro the streams to individuals, and that could actually be lowered by having individuals with spare bandwidth donate systems to act as leaf nodes. Just a thought for next time this becomes an issue :) But to spread out the asterisk boxes in theory you could support many hundreds if not thousands of clients at the same time off what appears to be the same feed. Using something liek ser (www.iptel.org/ser) as a front end you could provide a unified sip address to people and have ser do load balancing to the actual asterisk boxes acting as an application server. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection
I donÂ’t use app_amd but use waitforring to see if someone picked up, if not voicemail... Exten = s,1,wautforring(16) ; abiut 4 rings Exten = s,2,voicemail(1234) Waitforring waits upto N seconds if phone is ringing, if not ringing it returns -1. Dunno if this helps, but that works in my parents asterisk box with zap chan... -Original Message- From: Matt Florell[EMAIL PROTECTED] Sent: 10/6/05 5:48:09 PM To: Adam Robins[EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Answering Machine Detection If you have a copy that was released before it was wiped from existance would you be willing to post it for download or email it to me along with some description of how it works? Thanks, MATT--- On 10/6/05, Adam Robins [EMAIL PROTECTED] wrote: I just checked the 1.2 source. It looks like app_AMD is gone. All references to it on the Wiki are also gone. Can someone please tell me why AMD was removed? I am using it in 1.07 for several production applications. [Message truncated. Tap Edit-Mark for Download to get remaining portion.] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
Before people jump at the abstracts, remember that patent abstracts are very generic, and later they must be more specific. I am away this week but if its not answered by next I will look into it more, maybe there will be more news, maybe ill check pacer for the filing... -Original Message- From: Gleim, Jason[EMAIL PROTECTED] Sent: 10/5/05 2:05:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents I'll start with the disclaimer that I am not an attorney... nor do I play one on TV... But, a search of the US Patent Trademark Office reveals 13 patents assigned to Sprint that deal with VoIP. (http://www.uspto.gov/) 6947411 6944150 6937869 6909690 6870857 6868081 6865398 6741695 6731735 6697097 6681116 6556826 6373930 Of particular interest are the '9690, '4150, '1695, '3930 patents. '9690 is a patent on call admission control using silence suppression to better utilize network bandwidth. Specifically, it seems to deal with a method to apply adaptive silence suppression at the customer site... presumably in the ATA. '4150 is a patent on a 'gateway' layer to be implemented between a customer and the communications network as a means of offering and controlling services offered as well as optimizing the deliver of those services. '1695 is a patent on a method to interface packet-based and circuit-switched networks. It specifically mentions SIP and other protocols and how to interface them to signaling and voice paths in a circuit-switched network. Finally, '3930 is a patent on a method to 'redirect' call setup through a third party for the purposes of service restriction or authorization. Basically it's a method of implementing pre-paid service on a packet network. The only one that seems to me that would directly apply to the * community may be the '4150 or '1695 patents. But I don't know enough about patent law to know if it would be worth their time or if they would even have a case. There *maybe* something there too with some of the prepaid modules, like AstCC, if they could argue it was hosted on a separate system. Again, I don't know enough of the specifics to make an educated guess. OK... now that I did my part to add to the FUD, maybe somebody that knows more can build on what I found. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Wednesday, October 05, 2005 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-biz@lists.digium.com Subject: Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents At 2:43 PM -0700 10/4/05, trixter http://www.0xdecafbad.com wrote: Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge. Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. http://kansascity.bizjournals.com/kansascity/stories/2005/10/03/daily23 .html -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 This perhaps is quite relevant to the Asterisk community. While I don't know the specifics about Vonage, I do know that they have been rumored to have (in the past, or present) used Asterisk in their core for some services. (Voicemail? Conference? Messages?) This, however, is not confirmed. http://www.ilocus.com/ui_dataFiles/news18aug05.htm http://www.google.com/search?num=50hl=enlr=newwindow=1safe=offc2cof f=1q=%22vonage+uses+asterisk%22btnG=Search According to public information, Voiceglo uses IAX and Asterisk: http://lists.digium.com/pipermail/asterisk-users/2004-February/036311.ht ml http://www.business2.com/b2/web/articles/0,17863,1059204,00.html FYI: Voiceglo and theglobe.com are the same company for all intents and purposes. Therefore, I am very interested to see if this is merely co-incidental
Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
if the case is knowingly frivolous, vonage, voiceglo, and theglobe can sue sprint... Http://pacer.uscourts.gov should have the filing online, when I get home I may pull it unless someone beats me to it. -Original Message- My suspicion is that Sprint was in negotiations to acquire Vonage and they couldn't agree on a price so Sprint decided to sue Vonage to leverage their position. Regards, Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
I meant to imply anyone that wants to do business in america, not just live here, if you want to write off america fine, donÂ’t terminate calls here. The system is broke, and without money bigger companies can make life vary hard. My suggestion was to get more info, not to cease operations, find out specifically what is charged so you donÂ’t have to fight like voiceglo theglobe and yes vonage. Most people who want big business' will look to provide service everywhere rather than small regions only. -Original Message- Matt Riddell wrote: trixter http://www.0xdecafbad.com wrote: Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. Unless of course they don't live in the United Sue'ers of America. :D ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sprint Nextel sueing over VoIP patents
Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge. Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. http://kansascity.bizjournals.com/kansascity/stories/2005/10/03/daily23.html -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM tts engine integration
On Mon, 2005-10-03 at 16:56 +1000, Rod Bacon wrote: Not bad.. but still not as good as Scansoft's... do you have a url for an online demo? IBM's was just something I found that was easy to integrate into asterisk free. If scansoft also has a demo then I may look at writing something else to use theirs. I checked their website but didnt see an online demo. I am not happy with the lower selection in voices with ibm, but its free, simple to use, and works without any markers saying its a demo. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IBM tts engine integration
On Mon, 2005-10-03 at 08:56 +0100, razza wrote: trixter wrote: do you have a url for an online demo? http://www.scansoft.com/speechworks/realspeak/demo/default.asp Thanks its late so I prolly wont do this right now, but its a post method (same as sitepal and it looks easier than sitepal was). I will read their tos and make sure that anything I do wont violate that. 100 char limit it seems. Shouldnt be that hard, but I will be using netcat instead of wget/fetch. This does sound better.. if its usable the voice selection is also a lot more robust, and that is something I didnt like about the 2 choices with ibm (although my guess is that you can brute force voices out of ibm, its 3 letters in the url, I just didnt want to play those games incase they cried foul). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IBM tts engine integration
On Mon, 2005-10-03 at 08:56 +0100, razza wrote: trixter wrote: do you have a url for an online demo? http://www.scansoft.com/speechworks/realspeak/demo/default.asp I wont be coding this. It isnt hard if someone else wants to fine, I personally wont though. The reason is quite simple: This demo is for demonstration purposes only. For other use, please contact our Sales Office. I am not gonna violate something so plain :) However they dont appear to have much by way of security (although I didnt verify cookies I dont think they are using them in any way for this, and that is trivial ...). Its a simple form post, with variables for the language and voice as well as content. Anyone that understands basical HTTP should be able to figure out what to send, and how to save the resulting 8khz wav file, like with netcat for example, or perhaps a simple LWP perl script, and then use sox to convert, save and blah blah blah. perl would likely be easier than netcat, and most systems now require it (I avoid it because its not mandatory it be on a system, however I havent seen one without bourne shell). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] US tollfree DID request
I am requesting rates sent private to avoid list clutter for tollfree DID service in the US. please include instate vs out of state rates if different. Expect moderate to high volume on this account. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding Cepstral to Asterisk
On Mon, 2005-10-03 at 13:02 -0700, [EMAIL PROTECTED] wrote: the app_cepstral.c file had a problem that it was trying use #include ../asterisk.h I had to force it to where asterisk.h was located... in my case it was in /usr/src/asterisk/include so i changed the #include to say #include /usr/src/asterisk/include/asterisk.h and then it would compile through with no problems try adding -I/path/to/asterisk/includes in your case -I/usr/src/asterisk/include to your cc/gcc line (in Makefile usually CCOPTS var, but I havent looked at that Makefile specifically). This is the more elegant solution :P Or install your asterisk includes in the system default (/usr/include normally) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does the error stale nonce' mean?
A stale nonce is more of a warning than an error. In SIP your authorization credentials are encoded in the SIP headers. To prevent people from capturing that data and using it later to make calls on your account a nonce is used. A nonce is a disposable number that is added to the string a hash algorithm will hash. This makes hashing algorithms (like md5) have different output. This is a common cryptography technique. The SIP RFC requires that the nonce randomly change periodically. If the client uses a nonce that was expired it is considered a 'stale nonce'. The client should then get the current nonce and use that instead. This message lets you know that the client tried to use a stale nonce, which can indicate someone trying a replay attack (using captured data from a previous session) or a client that isnt properly getting the new nonce, or even just timing issues as follows: Client gets a nonce. Client goes to register/reregister using that nonce At the same time the client is preparing the message to register/reregister the server chooses a new nonce Client sends the message with the now old nonce Then again it could be something else entirely :) On Mon, 2005-10-03 at 22:35 +0200, Morten Isaksen wrote: On 10/3/05, Olle E. Johansson [EMAIL PROTECTED] wrote: Does anyone know what stale nonce is? I've answered this question many times, so you should be able to find the answer... A stale nonce is when a device tries to re-authenticate with a nonce that is no longer valid. We are telling them that the nonce they used is invalid, and re-issue a new challenge and a fresh nonce. It's just an informative message, that I propably should move away to a debug level of some kind. I get this error when I use a Audiocodes MP-124 against Asterisk 1.2beta1 and asterisk refuses the call. When I use CVS-D2005.02.12.14.37.11-04/13/05-16:14:03 it works fine. I do not have access to the debug and log file now, but I will send them tomorrow. /Morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on windows
On Mon, 2005-10-03 at 17:27 -0400, Paul wrote: As for X on the same box as *, it only seems to affect calls when I do something that uses enough cpu. I can be logged in with a gnome or kde desktop without causing problems. It's a P4 2.4 with 1 gb DDR 333. For smaller volumes of calls (10-20 concurrent) I havent had problems with call quality while running X, and many X apps with a AMD 3200+ (1.4GHz) and 512MB ram. You can go fairly low end and still run X as long as you dont run an X operating system (ie one of the window managers that takes 500MB ram by itself). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM tts engine integration
I wrote a very very simple shell script and an even simplier macro to use the IBM TTS engine within asterisk for prompts. While its free you are limited on the number of requests you can do within a day. If anyone is interested its available at http://www.0xdecafbad.com/Asterisk-Text-to-Speech.html -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can an outside caller dial an extension before someone answer?
Short of making this time based or having multiple inbound numbers you cant do this without answering the call and reading dtmf (or as explained this last week T1/E1 lines may or may not be able to pass audio data incl dtmf for upto 90 seconds when it starts to ring). Now here is a problem, how would someone know to dial the extension if there is no automated attendant telling them to do so? Most callers wouldnt know to do this as most systems dont accept this. There is a trick for those 'in the know', asterisk answers the call but instead of playing a recorded voice it plays a ringing tone. Users would then be able to dial an extension if they know to do this. On Mon, 2005-09-26 at 16:48 +1200, Simon Glass wrote: Hi, We don't want a digital receptionist if we can help it (too impersonal!), but is it possible for an outside caller to dial an internal extension (eg 201) after asterisk answers the call, but before someone in the incoming call ring group has answered? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] change codec based on callerid (sip/iax)
I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser (www.iptel.org/ser) in between the asterisk box and forward effectivly to a different account on the asterisk box based on caller id (ie ser makes a choice which account to use). codecs then would be negotiated normally at connect time. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context question
That doesnt really help. As stated in the email you replied to what is to prevent someone doing say [1] exten = 1,1,goto(2,1,1) or customer A *and* customer B trying to define the same context name, to use your example lets say they both want to create context '1'. I want to be able to create 1 system that has multiple users who are able to create their own dialplans without naming collisions with other customers or gotos going to other customers, etc. This is more for a virtual hosting type setup so I can have one large machine instead of many smaller ones, thus allowing for better ROI. While many have suggested that I learn the basics of contexts (as you did) no one has been able to ansewr the actual question asked making me think there is no current answer, and an AGI is the way to go. That way I can have more control over what data is observed and all that. I just didnt want to write an AGI if there was an existing solution, especially if it was part of asterisk itself and not an external program. On Mon, 2005-09-26 at 09:31 +0200, Bruno De Luca wrote: this can help u: EXTENSIONS.CONF [1] exten = 1,1,Dial(SIP/1) exten = 3,1,Dial(SIP/3) [2] exten = 2,1,Dial(SIP/2) exten = 4,1,Dial(SIP/4) trixter http://www.0xdecafbad.com wrote: They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip, call ransfer and call waiting
On Mon, 2005-09-26 at 11:08 +0200, Daniel ANDRE wrote: Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on conversation (off hook). But I don't want to loose call transfer, call hold and so on. Is there any possibility to do that? Yup... exten = 123,1,SetGroup(user1) exten = 123,2,CheckGroup(1) ; dont let more than 1 call at a time exten = 123,3,Dial(sip/user1) exten = 123,103,Busy ; this is where it goes if CheckGroup indicates more than X calls ... see http://voip-info.org/wiki-Asterisk+cmd+SetGroup for more info. You may have to play games with variables to make a macro perhaps that would be more generic in this regard, but this should at least get you started. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] change codec based on callerid (sip/iax)
On Mon, 2005-09-26 at 10:33 -0700, Michael D Schelin wrote: This can be done by modifying the source code. how helpful. If I modify it enough it will be 100% identical to windows xp, anything can be done by modifying any code. That however doesnt answer my question with anything that isnt obvious, such as is there a way to do it without a modification? Would the ser idea work (which may be better as the call volume would likely exceed asterisks ability to process calls anyway? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialplan game
On Sun, 2005-09-25 at 08:58 +0300, Tzafrir Cohen wrote: On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com wrote: Has anyone built a game with the dialplan? I would think this would most easily be managed by an AGI, but its possible with realtime extensions. The game would be like 'adventure' that I first played on a prime in 1979. Or any of the infocom games (ie zork). Infact since the infocom spec is known it might be possible to plug in the data files directly from an AGI. If anyone has done this I would love to hear about it. Such a game requires the player to keep a lot of state information in the head. Why not start with something simpler? Becuase something else is not what I want. The ability to read dtmf and play some response via a TTS engine isnt that hard, just wanted to talk to someone who has done it, specifically to get the scope they did and some other info. And you dont have to keep a lot in your head, there isnt a lot you can do with a keyboard and monitor, so why do you think you have to keep a lot of information in your head? Also, looking at the package bsdgames, some games are command-line based and thus could be adapted to a dialplan control. There are some adventure-type games. And there is also monop (monopoly). Though frankly, I'm not sure those would be of any atraction to any user. I dont quite think you understand what I wanted. Monopoly was not a text based adventure game like zork or adventure was way back when). I am not looking for any game that can use text as its output, I am looking for text based adventure, possibly MUD to allow for interaction with others (although that can be slightly more tricky). You do need the game to sing a little bit, as it can't dance. But singing will become annoying after a while if there's no simple way of skipping it. what? All it needs is a quality TTS engine and the ability to read dtmf. There are alternatives to festival so the TTS end is covered, and unless something fundamentally changed it can read dtmf with the greatest of ease. So I dont know what you are talking about with relation to singing. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialplan game
On Sat, 2005-09-24 at 23:30 -0700, trixter http://www.0xdecafbad.com wrote: On Sun, 2005-09-25 at 08:58 +0300, Tzafrir Cohen wrote: On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com wrote: Has anyone built a game with the dialplan? I would think this would most easily be managed by an AGI, but its possible with realtime extensions. I found one not yet finished example if anyone else is interested in this, http://uc.org/read/Zasterisk it uses the infocom worlds (data files that describe each room in the game). I am still interested in others that people have done, and would like to speak to the developers of those apps. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Voip provider
On Sun, 2005-09-25 at 06:20 -0400, Nana Tandoh wrote: Termilink Digital Voice www.termilink.net On 8/13/05, jonny hashem [EMAIL PROTECTED] wrote: what is the best voip provider that provides good service ,good voice quality and good rates . any one have an experience with voip providers advice me. How do you define good service? tech support or voip service? Good quality and rates to where? A provider that may be cheap to one place is more expensive than others to other places. A provider that has good quality calls to one country may not to another. Further do you want sip, iax, something else? Or do you not care? Selecting one protocol over others can reduce the list somewhat (although most provide sip some are iax only for example). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Voip provider
On Sun, 2005-09-25 at 14:37 -0400, Leif Madsen wrote: Its http://www.mixnetworks.com - not .net. Sorry! They dont seem to have their rate information easily locatable, and I am afraid of voip companies that hide their rates. They have a pdf (I get only a blank page) for the 'local calling area' or some such ... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] context question
On Sat, 2005-09-24 at 09:10 -0400, Alex Vishnev wrote: I briefly looked thru the code and I don't believe there is a way to separate the context or really make them independent. I know exactly what you want to accomplish. I think it could be done with a little trick. For example, every customer on hosted pbx would be given some kind of unique identifier. The back-end would silently place the identifier at the beginning or the end of the context making the new name totally unique. The front-end would hide identifier from users view and just present the name of the context. That way, customers can name their context anything they like and there would be no collision. In that case, Goto would also be local to the context as the real context name will contain customer id. Does that work for you? no, because as I stated I didnt like that for personal reasons. That sounds exactly what I was thyinking too, prepending some customer specific identifier. If that is the only way to do this, then I think I will just have to run everything through an AGI, which can differentiate between customers since none of the 'dialplan' is in extensions.conf :) Thanks though, at least its confirmed that this doesnt exist (yet anyway). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialplan game
Has anyone built a game with the dialplan? I would think this would most easily be managed by an AGI, but its possible with realtime extensions. The game would be like 'adventure' that I first played on a prime in 1979. Or any of the infocom games (ie zork). Infact since the infocom spec is known it might be possible to plug in the data files directly from an AGI. If anyone has done this I would love to hear about it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context question
Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context question
They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] custom ring tone
On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote: yes, yes the thing is that local telco uses this feature for their customer support line and also one of wireless providers now also offers ability to customize your ring tone I was told that if you have analog or even ISDN BRI line that ring tone is generated in your local teclo exchange, but if you have connection like E1 that it is generated localy in your PBX (explanation being that So in short you can have a toll free info line without actually paying for the toll free. While its not interactive, by not sending answering supervision the caller is not charged. Interesting concept they have there, sure beats the 10k resistor trick from the analog switch days (although then you could talk to the other person). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] custom ring tone
On Thu, 2005-09-22 at 14:02 -0600, Colin Anderson wrote: ?? Ringback is provided by your PSTN provider until answer by asterisk. You have no control until you answer Then you go to IVR, VM or ?? OP said T1/E1, so (usually) there's no ringing delay until Asterisk picks up like in a POTS line. Answering a T1/PRI line is transparent to the caller and then you can fake any ringtone you want. So: Check your local laws on that, in America there was a telephone company many many years ago that got into trouble for doing that type of stuff as the caller was billed but got what they thought was a ring or busy but never an answer. I dont know exactly which state or what set of laws the telco was found to have violated, but the call was not answered technically yet answering supervision was sent. The laws may have changed by now or been obsoleted by new ones I have no idea. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] custom ring tone
Audio both ways? Sure would beat the collect call game :P On Thu, 2005-09-22 at 21:15 -0400, Alex Vishnev wrote: Actually that is not true. You can have a short time where audio path is open prior to answering of the call. This depends on the provider, switch and software. I think the largest window I have seen is 90 seconds. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Thursday, September 22, 2005 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] custom ring tone On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote: yes, yes the thing is that local telco uses this feature for their customer support line and also one of wireless providers now also offers ability to customize your ring tone I was told that if you have analog or even ISDN BRI line that ring tone is generated in your local teclo exchange, but if you have connection like E1 that it is generated localy in your PBX (explanation being that So in short you can have a toll free info line without actually paying for the toll free. While its not interactive, by not sending answering supervision the caller is not charged. Interesting concept they have there, sure beats the 10k resistor trick from the analog switch days (although then you could talk to the other person). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording
On Wed, 2005-09-21 at 10:07 +0300, Zoa wrote: The reason i recommended you to use a ramdisk is because i think the problem with recording to disk is saving 20ms of stream 1, then 20 ms of stream 2, then 20ms of stream 3 etc etc meaning you write everytime very small things. (with a lot of seeking). Our best test results were with: filesystems are also a consideration with larger scale projects. Different filesystems add different amounts of overheads on different types of operations. Some are faster at moving small files around others faster with large files. This adds to the disk latency. Removing the disk latency itself is a good thing, since that is typically slower, but to crank out that last little bit of performance some research into the different filesystems under the specific kernel that you are using could also be a consideration. The most obvious (and easiest to update a running system) is to remove things like atime, whih with most linux distros is on by default. This causes a write operation for the read of a file to update the last time accessed. A couple little things can add up to a few percent improvement and generally make the cost go down. - buffering the recordings to a ramdisk, then - on low load (at night) copy the files over the network (easy to shape the pipe, so that you dont overload anything), Or have a seperate network set up (dual nic card for example) where the 2nd network is used just for NFS traffic. Although NFS generally is ugly network wise, it is standard and makes things easier. Just gotta watch the IO on the system given that the network card itself will cause cpu cycles to be used, but lets face it cpu is cheap now. Different drivers also work differently, and then with the 2.6 series kernels you can use device polling instead of interupts which can help a little. If you want to go even freakier, run asterisk (or you complete distro) from a ramdisk. When you say ramdisk here I assume you mean using conventional ram, its cheap yes but its volatile, do you have any plans for failure of the system or ram? Or is the data integrity itself not as critical? The reason that people like hard drives is because most of the time if the system goes down for any reason the data is still intact. I thought over your suggestion to use a sniffer to do the recordings, you might pull it off, but will have to write your own to do so. (or go to the expensive version of commercial sniffer applications). isnt vomit free? It was a voip sniffer that worked with some codecs many years ago (I wanna say mid-late 90s but I may be thinking of another back then). http://vomit.xtdnet.nl/ does G.711 only. The bigger prIoblem that I see is that sniffers dont always get all the traffic that is on a network particularly when the network has more traffic on it. While this generally isnt a concern and I would like to think that even a poorly configured network could allow for 512 calls, it is a factor to implement this type of a solution. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording
On Wed, 2005-09-21 at 11:11 +0300, Zoa wrote: Also when you do things over the network, disable your onboard network card, and go for some more expensive network card. In our tests with small packets, we could increase the throughput with a factor 2. (related to cpu load). I wonder how much of that is a poorly written driver and not the card itself. I have seen some fairly poor drivers performance wise. :/ -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ftp.soft-switch.org down?
On Wed, 2005-09-21 at 20:13 -0500, Anton Krall wrote: Guys, is Steve's ftp site down? DNS says ftp.soft-switch.org doesn't exist. Anybody else seen this? Its happening here. I checked a few things, domain is not expired, joker DNS is serving this domain and its up. www works, so it is a host specific entry. The www IP is a godaddy IP, so it appears that its hosted there. In short steve has to comment on this one. Maybe the server is getting moved? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HooDaHek 0.6 Released
On Tue, 2005-09-20 at 14:31 -0400, Jonathan k. Creasy wrote: Yellowpages.com has a reverse lookup on it. http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp As does whitepages: http://www.whitepages.com/10001/reverse_phone http://directory.google.com/Top/Reference/Directories/Address_and_Phone_Numbers/ and lets not forget google itself (residential only aparently) phonebook:QUERY (smith, ca or 2025551212) There are a lot of them out there, used by stalkers every day. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ${DIALSTATUS} problems
On Wed, 2005-09-21 at 14:36 +1000, Mark Edwards wrote: terminating asterices. (Is that the plural of asterisk?) I propose asterii, while by no means gramatically correct it wont fall under potential sue happy lawyers who own the unix trademark (after all the plural there is unices). oh no I said unix and didnt credit anyone or pay royalties. They are gonna get me now. :P -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kill a .call file
On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote: Any means of killing a .call file that is in progress? You mean once the call has begun? You prolly want to hangup the call ... asterisk -rx soft hangup callid Or is there something else that you wanted? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Differ between private and out of area?
On Mon, 2005-09-19 at 13:48 -0500, jltaylor wrote: out-of-area is displayed for calls that originate from LECs that have not implemented caller id. Or for companies that dont share it, this is sometimes the issue for foreign originated calls. Caller id is sent via SS7, and some companies that do have caller id, and do have SS7 for other aspects for some reason do not transfer it globally. I have seen this from calls from the UK to the US for example (but not all such calls). In America the FCC basically requires that if you have caller id support you must pass caller id data, so most companies in the US pass caller id. The federal government however always seems to pass 000-000-, I guess to keep it 'private' but not trigger any privacy blockers so the call goes through. MCI sales team used to pass 'out of area' for the same reason. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting call minutes on a GSM SIM
On Wed, 2005-09-14 at 09:30 +0200, Remco Barende wrote: Thanks for the tip. I was actually thinking in the direction of putting the asterisk calling card application to use. I've never used it and wonder if it is at all possible to use it from within the dial plan instead of normally from an extension. Yup. I will try to make it simple for the archives, or anyone else that is interested in doing this type of thing. You appear to know most of this already, but then again you arent the only person on this list :) Call the AGI from the dialplan when you want to. exten = 31337,1,answer exten = 31337,2,playback(welcome) exten = 31337,3,agi(blah.pl) replace blah.pl with whatever the name is, so long as its executable. blah.php blah a.out etc see asterisk.conf for where to place the agis astagidir = /some/path/to/asterisk/agi-bin -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting call minutes on a GSM SIM
On Wed, 2005-09-14 at 07:01 +0200, Remco Barende wrote: Hi! I'm considering to buy a GSM bridge to save on GSM calls. Right now they are offering subscriptions with 200 minutes each month for almost nothing, however the 400 minutes subscriptions are considerably more expensive. Most GSM bridges can cater for 2 SIM cards, is there a way for Asterisk to run the first SIM card to it's max and then switch to the second? (If one call would overlap I wouldn't mind). Asterisk would have to keep track of the minutes called each month for a SIM (channel?). On most bridges you can select the SIM you want by a dial prefix. I do not know about the specifics, but it seems to me that you would need an AGI that would track the usage and compare that before placing a call. To switch I do not know how you tell the sim adapter which one to use, but surely there must be a command somewhere, the mere fact that agi allows you to script something like this fairly easily means that it shouldnt be a big problem, assuming you code :) And you can even pick your favourite language given how the AGI talks to asterisk even 'unsupported' languages can be used. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] motorola vt1000 games
For those that are interested in the vt1000 paper I wrote a while back, I have it now on my webpage, at http://www.0xdecafbad.com/Unlocking-Motorola-VT1000.html Some of the information there was posted elsewhere, some wasnt. basically the unit runs vxworks, and it needs a docsis like server to reconfigure properly (its different from the cable modem docsis stuff mot does) but ... I think most if not all the hardware is supported in linux, I didnt really check just glanced at the chips, it may be possible to reflash the unit with an embedded linux version and run a very stripped down asterisk implementation, thus making the units more valuable, and since there are many in surplus now with no large provider supporting them anymore, you may be able to get em really cheap (I believe ebay has some for cheap). There is support to reflash without difficulty, providing you use the vxworks boot loader, there is a connector that may be jtag which would let you more easily reflash without doing dev work under vxworks to write a loader app on the unit. Again this is stuff I didnt really look into. Downside is that JTAG is slow so you prolly wouldnt want to reflash the whole thing via JTAG but just enough to get to the serial port and read the rest of the data that way. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Good Deal on A Good Asterisk Box?
On Sun, 2005-08-28 at 21:05 -0400, asterisk wrote: I am assuming two, couldn't a USB NIC be used? Obviously not gigabit but can anyone see any problems with that setup? USB throughput is less than max bandwidth which is what is advertised. Add a hub and it gets even worse. There is a substantial framing overhead for usb. USB 1.1 has a raw transfer rate of 12Mbps USB 2.0 has a raw transfer speed of 480Mbps I believe the polling of USB devices is slightly more processor intensive than of a pci card, but could be wrong (and then it may just be the drivers that make it appear that way). In theory 100Mbps wont have a problem on a USB 2.0 host, plenty of bandwidth to spare, and depending on application it may be acceptable. I however would not use such a device in a busy data center/colo for fear that someone might unplug it (accidentally or intentionally) since usb doesnt really lock in place. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd
On Mon, 2005-08-29 at 11:37 +1200, Matt Riddell wrote: cmisip wrote: I want to be able to send a dtmf key to asterisk and have mplayer forward or rewind. pabx*CLI show application ControlPlayback mplayer has advantages of more codecs as well, so you arent as limited. In addition it will play tv (with tuner card), dvds, etc. So you can really pick what you want as your audio source. It would seem to me to be not that difficult with an agi to use mplayer, although I havent tried. controlplayback seems to fit if all you want is mp3s however ... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd
On Mon, 2005-08-29 at 15:24 +1200, Matt Riddell wrote: trixter http://www.0xdecafbad.com wrote: controlplayback seems to fit if all you want is mp3s however ... Although it works with all supported formats. how many are supported? mplayer for example does at least 130 codecs making it easier to get whatever you happen to have. That was my point. You always could use mencoder or other tool to convert to the desired format, but that isnt always an option (usually it is though). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Satellite Broadband and VOIP
On Sat, 2005-08-27 at 12:14 -0700, Julius Igugu wrote: I use a satellite connection and VOIP is ok! It depends,mostly, on what you expect! There's an inherent delay in the system usually about 700ms - 800ms! but this is bearable. That depends on how you define satellite and the network itself... Satellites are things that orbit the earth (duh) but where they orbit is critical to latency. Some systems are in geostationary orbit and have much higher latencies. Radio signals go about the speed of light (just to cut off anyone who wants to correct my 'about' comment ... when passing through matter even light slows down). LEO (low earth orbiting) satellites that are not fixed to a given location in the sky are much closer, and have much ess latency. The newer variant of artificial satelites is a solar powered airplane, which is WAY closer and thus has much less latency (about 18 miles so the latency is hardly detectable from the RF link). afaik there are no stratelites (what the solar powered airplanes are now being called because they fly in the stratosphere) in active deployment but there are companies planning on doing those. 1 stratelite can do an area roughly the size of texas (for non americans that is a very large area, google it if you want to know the sq km/mi :) At 18 miles you still need a good antenna system (wifi has done 125 miles at 11mbps unamplified with a 300mw card so it is possible ...) A lot of satelite providers overload their networks for cost reasons (its really expensive to upgrade the routing equipment) as such those devices add extra latency to the mix, geostationary orbit is about 250ms each direction so anything over 500ms is normally caused by routing equipment not being able to transmit as quickly as it should ... With the newer systems coming out it should be even better. 18 miles would be no worse than wire based systems in most instances, provided the equipment is upgraded regularly, and time to deploy, dost to upgrade, etc are lower so that just might happen. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] US based CLEC provider request
I have a proposition for US based CLEC(s) and would like to speak with any that read this list offline. In short I am looking for US DIDs for high volume traffic. If there are any CLECs out there, please contact me offline via email. Thank you, Bret McDanel [EMAIL PROTECTED] -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Vonage locked Motorola VT-1000s
On Tue, 2005-08-23 at 15:16 -0400, Douglas Logan wrote: That username password combination is referenced elsewhere for different models of ATA's as well. I believe it is somewhat a Vonage standard. one of the things about the vt1000 is that the provider can dynamically change your pw. That is part of the file that gets tftped to the box at startup and periodically after that. I dont know if vonage does this, but they had the ability to do it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Vonage locked Motorola VT-1000s
Steve Gladden wrote: I have a small pile of them from customers who were too lazy to send them back after switching to our local voice service... Is there any hope of ever using these things with Asterisk? Vonage does not want them back and they won't unlock them either. A terrible shame! Should I just toss them? Steve I wrote a paper on how to 'unlock' them, the short is that without a mot server (similar to the cable modem docsis stuffs) you cant do anything highly meaningful with them. I hope to have my webpage back up soon (it was being physically moved and the people that are doing that broke some stuff in the process, but hey its free). You can see what I did and maybe take it from there. There is a TTL serial port inside the case, I used a TTL-RS232 converter and connected to it, it runs vxworks, and I mapped out the urls that are valid (incl the 2 undocumented ones) and some of the memory addresses the profile info is stored. All I can say is that if you are highly interested in this check my page occasionally over hte next little while, I couldnt find any of this on the net anywhere, maybe google cache has it. http://www.0xdecafbad.com/ I checked while writing this email and the vast majority that was on my site is not cached right now :( -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 ___ 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] Vonage locked Motorola VT-1000s
Steve Gladden wrote: Very Highly Internested Any chance you could zip or tar your content up and email it to me or give me a link to grab it? Maybe I could help you get it hosted again too ifyou need that. Thanks!!! Steve I would love to have a tarball of my web stuff. I didnt know it was getting moved, and it got moved earlier than expected. I will see if I can get a tarball myself (I should have kept my own backups but ...) -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.13/78 - Release Date: 8/19/2005 ___ 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] Vonage locked Motorola VT-1000s
Steve Gladden wrote: Well hey! let me know!!! :-) I got my max232 chip sitting out and am building a converter board right now... Gonna give it a shot soon as I get yer info!!! :-) Have you done successful re-blast on one of these before? Very familiar (well kinda) with Motorola vxworks surboards etc. Take care! Steve I briefly looked at hte hardware and it *appears* that linux supports all of it, or at least enough of it that it shouldnt be terrible to port linux (embedded) to this device. I did not fully look into this, it was more like 'that realtek chip looks supported' rather than pulling specific model numbers. I also do not recall any of the hardware now, but ... I am told that the server is in the mail and should arrive at the destination soon where (hopefully) no data was lost (those pesky bits love to fall out of the seams of the box) and it should be up soon. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.13/78 - Release Date: 8/19/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: RE: Business Edition
On Fri, 2005-07-22 at 21:26 +0100, Kevin Walsh wrote: Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 22 July 2005 12:04, Lee Howard wrote: Well, I'm sure that was an added bonus. :-) Free work and free money. It reminds me of a certain Dire Straits lyric. Yes but are the chicks free? All except for the binary ones. binary only chicks are ready to be executed without a lot of configuration or compilation so that may be a good thing :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] a ne pas voir
On Thu, 2005-07-21 at 10:41 +, ali kia wrote: hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group If that serves you better than this list or the existing irc channel (irc.freenode.net #asterisk) then by all means go for it, however I think you may find that getting a massive group to migrate to something new will be difficult. You may find that it is easier to use irc for real time chat and this list for email queries just because that is where everyone else already is. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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: Re: [Asterisk-Users] a ne pas voir
On Thu, 2005-07-21 at 15:56 +, ali kia wrote: you can download amsn it work under linux i have it and it works succesfully I think he was refering to the service provider, MSN as in MicroSoft Network, as opposed to the operating system. There is already a large enough user base and comments about the irc channel irc.freenode.net #asterisk, which is accessable via many clients in many operating systems and even via web browsers if you have a server with the appropriate software in place, or applets local to your system. To make the MSN chat meaningful you would have to get a bunch of people to convert, and it is always much harder to get everyone to change the way they currently do things to something new unless you can prove that the new way is somehow superior. MSN does not appear to be superior to any other realtime chat network, so it will be a tough sell. You can obviously go there yourself, and attempt to get others to follow you, I just see that path as a difficult one, the easiest one would be to follow the crowd and do what they are already doing, but innovators never got anywhere by following the crowd. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
On Thu, 2005-07-21 at 10:19 -0700, Lee Howard wrote: What I am saying, though, is that Digium didn't give out royalty-free proprietary licenses to Asterisk, instead, they gave out GPL licenses to Asterisk. Why, then, do they require that contributions are made any differently? Why do they require freedoms with contribution that they did not give with theirs? Well, probably because they believe that they're owed that, and probably because many others in the community not unlike yourself agree with that opinion as well. There was some discussion about a month or so ago, and a digium rep piped up to even help try to clarify this particular issue if memory serves. You do not give up your copyright on your contributed code. You do not have to give them full rights to your code if you do not wish to. You have an option to contribute GPL only code. If this is incorrect I fully expect a digium employee to speak up, but if I remember correctly that is what was said at the end of the previous thread. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VoIP provider
On Tue, 2005-07-19 at 07:12 -0400, Chris Mason (Lists) wrote: Madhawa Jayanath wrote: o Bernie, 1) best results www.nufone.net 2) low cost www.voipjet.com Anyone able to find NuFone's rates? I have been looking for them on their site. I need international rates and UK Mobile. As there is still 182 emails I havent read yet this may be answered (that will teach me to leave for a day). http://www.nufone.com/rates.txt iirc. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VoIP provider
On Tue, 2005-07-19 at 12:42 -0700, Derek Whitten wrote: rofl.. nufone sends you configuration information via email after you sign up for an account.. On Tue, 2005-07-19 at 11:16, Andrew Kohlsmith wrote: Nufone seems to have always been a DIY type of VOIP provider. Their new members page works very well and shows connection information and so on... maybe their email was blacklisted by some spam filter on your side? They dont do much marketing on their page, rates are hidden from normal view (ie nothing on the main page that indicates 'click here for rates'. This indicates to me that their advertising model is largely word of mouth, and it seems to be working for them. The fact that they rely on word of mouth advertising and are apparently doing well, that speaks volumes considering the fact that so many providers have had problems at some point and yet people keep recommending nufone. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Hottie ?!?
On Wed, 2005-07-20 at 21:02 -0700, Steve Totaro wrote: I have connected to a 3com system via t1 and e1. t1 and e1 are standards, not proprietary. Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports EM Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does the phone systems have to be the same ? I think the confusion comes in from the different ways that T1s carry data. In general yes you can connect a traditional pbx to asterisk via a T1/E1/J1 card providing they both can speak the same dialect. Typically vendors make their interfaces support everything because you never know what variant they will connect to. It *might* use some secret proprietary standard on the pbx side but I would say if it does that would be exceptionally rare (I havent come across one that only does some secret propritary format). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VoIP provider
On Tue, 2005-07-19 at 02:02 -0400, Bernie Courtney wrote: looking at setting up an asterisk box at my home-- what VOIP providers are you all using with the best results (and low costs! lol) thanks Bernie That is a hard question based on what you have stated. First if you are using asterisk that limits your choices, not all appreciate it when you connect something they didnt give you. As for best that depends, sometimes one is better simply because network wise its closer to you, other times its because of the level of tech support, generally its because of call quality and percentage of dropped calls. You mentioned low costs, but that depends on how you will use it. For inbound only? outbound to which countries? small minute usage or high volume calling? It all depends on how you plan on using it. You can also mix and match. For inbound only with a washington state US number ipkall.com is good and free (they forward through freeworlddialup.com also free. For outbound you can use nuphone.com or something like that that only does outbound. That way your costs are only when you actually place a call as opposed to whether or not you do. It depends highly on how you would use it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] So you all think VoIP sypply is warm and fuzzy
On Mon, 2005-07-18 at 17:38 -0500, Brian Capouch wrote: I'm not positive that a vituperative letter like this sent to the list unsolicited is going to win you a lot of support in your crusade. given that he said he would only do this if they didnt give him the CD (or strongly implied it anyway) that makes me curious as to why it was sent before any resolution could be had. Its one thing to try to resolve issues with the vendor but to publicly try to attack them in a forum where its unlikely they will respond (based on historical posts about them and their lack of comments from the company themselves) but pretending to give them a chance to make it right ... I dunno. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 licensing - Hardware Devices rather than software
On Tue, 2005-07-19 at 00:35 +, Obelix wrote: I have been reading a number of the past threads about G.729 licensing., about how the registration keys are linked to the network configurations, limited number of registrations etc, etc. Is there no reason why the decoding can't be done in with some Asterisk compatible hardware, so that once the adapter is bought, all licensing issues go away. In that way the owner could fiddle with the installation to his hearts content, without having to bother about reregistering licenses after some changes. It would save both Digium and end users a lot of hassle. They need to ensure that the license is not used by others. Digium has to pay the patent owner a fee for the codec. The way that it is licensed by the patent owner is per concurrent use as well. In linux gethostid() returns the IP address, not all systems work this way, some use a serial number off an eeprom (sparcs for example). Without locking it to something hardware based (cpu serial or something which isnt guaranteed to be accurate since its trivial to make a sysctl to report whatever you want ...) that woud be a feat. Additionally if you lock it to a peice of hardware you would not be able to play with the hardware, only the network. gethostid() is a silly way to lock hardware in my opinion anyway since it returns the IP address and many people now use NAT (by need or desire such as perception of increased security). NAT allows the system to sit behind the real IP and dish out seats and its possible (although it would take an illegal act on all concerned parties) to use the software without actually paying for it (someone somewhere would have to pay for it, but ...) Additionally with LD_PRELOAD or programs like systrace (depending on how its done in the code) you can force gethostid() to return whatever arbitrary data you wanted on a per invocation basis. One program can get the hostid as X while another on the same system at the same time gets it as Y. But right now this is the best of everything because it does not force you to buy additional hardware you may not have and do not want. And unless the communication path to the device could be controlled or a crypto system was implemented (and ITAR may be a problem, although I think they have exceptions for devices like this) the hardware could be emulated via software and it would totally defeat the licensing system with about the same degree of ease. All it would do is add cost to the end user, something I am sure most people do not want. In theory asterisk could bridge the licensed codec to an external hardware device that would have the number of seats in it but this would add latency and degrade performance, something I am very certain people do not want. What exists is the best of all worlds given the world we live in. Patents do exist in some places and as such the patent holder has the right under those laws to charge if they desire. In this case they do desire, and so digium is forced to pay. Being responsible business people they pass that charge on to the end users as it would be foolish for them to asorb the cost so that everyone else does not have to pay. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Concurrent users
On Mon, 2005-07-18 at 21:46 -0300, Rodrigo Otavio de Fraga wrote: How can I calculate the quantity of concurrent users using a bandwith of 512Kbps ? All users are using G.729 codec. How large is each packet? There is packet overhead for each packet that needs to be counted in (at the very least ip/udp but likely link layer framing and rtp data that comes out of that as well). The codec is roughly 8kbps so you can in theory get 64 channels, but it will never ever work that way (on the net) because of the packet overhead. Odds are you would be safe with a guess of 40-50 but I havent sat down and done the math because I am not going to guess on your sample size and such. I think that the wiki needs to have (if it doesnt already) explanations of all of this along with jitter buffers, what they are, why they are needed in some configurations, etc, since that does play into this from a performance angle. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 licensing - Hardware Devices rather than software
On Mon, 2005-07-18 at 22:19 -0500, Kristian Kielhofner wrote: While I do appreciate the lesson in system calls, what does any of this have to do with the g729 codec? :) Digium's G729 codec (and registration program) binds your license key to the MAC addresses of the ethernet adapters in the system. Even then you can register to three different sets of MAC addresses before you have to contact Digium to have your key reset. What do IP addresses have to do with anything? if you read what I said you would understand why I referenced IP addresses, which I assumed, aparently incorrectly that is the call they used (becuase it has been a standard call for licensing for at least 2 decades). As for the mac address ok that just makes it simplier to deal with since the app doesnt directly interface with the hardware (the kernel does) it is trivial to set the mac addr to whatevr you want (ifconfig does this with many drivers) which makes it an even more moot point than having to have code to 'play' as the original author wanted. Just gotta watch that you dont have two with the same mac addr in some networks (some systems and network devices dont care enough others completly come unglued). Wow. Anyways... good response for someone that had to be explained yet again why I said what I said because of their inability to read in the first place. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 licensing - Hardware Devices rather than software
On Mon, 2005-07-18 at 21:45 -0600, Tim Pushor wrote: Just gotta watch that you dont have two with the same mac addr in some networks (some systems and network devices dont care enough others completly come unglued). Yeah, like ethernet. let me clarify, on an ethernet network some systems and devices dont care others freak out. happy? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 licensing
On Mon, 2005-07-18 at 23:45 -0500, Kristian Kielhofner wrote: Ethernet (specifically mentioned over and over here) does NOT handle duplicate MAC addresses very well. At the very minimum, you would knock at least one of your cloned Asterisk machines off of the network, pretty much defeating the purpose of the scam in the first place. not totally true, it is on paper, but implementation doesnt always mean the same thing as paper. I have had multiple systems with the same mac address on the same network for other purposes and did not have problems with those systems talking to each other or other devices on the network as a general rule. Same MAC different IP. I have seen some switches freak out becuase the same mac addr is on multiple ports and it doesnt know which one to send it to. The original thing I said that resulted in 'wow anyway' as the most clever response I have seen in a while did include methods that would have enabled one to defeat the MAC address checking without actually changing the MAC address, again proving that copy protection is largely a waste of time from a technical standpoint, but not one from a business standpoint. Personally I dont think that digium would care if everyone had G729 but the patent holder does, so they must respect those wishes. As for what digium was going for I think they were doing what they had to so they wont be sued since they dont own the patent and have to pay someone and we live in a sue happy society. No matter what they did there would be a way of defeating it, some just require less skill to defeat. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Validating a phone number
On Sun, 2005-07-17 at 00:01 -0700, Peter Hsu wrote: I'm concerned about people dialing out of our asterisk server to numbers they shouldn't be dialing. Is there a concrete algorithm for determining whether a phone number is normal. i.e. calling this phone number would result in a normal long distance rate. It seems like the pattern 1NXXNXX for the U.S. is fairly commonly used, but it wouldn't catch erroneous phone numbers such as 1411XXX (and the other X11 numbers) normally those dont work with a 1 before them, but I cant say that is a guarantee with all providers. 19xx is normally premium service and has a sometimes steep charge. 1700xxx is another number that normally places a higher than normal charge to callers for calling. In america there are some numbers that appear normal but are premium service numbers, there are some in NJ that charge $5 to call in the 201 area code but they can exist in other states as well. You may want to filter numbers that would fit the 1NXX... format but arent in the US or Canada either. There was a company that had a number )I forget where somewhere in the caribean) that was part of the NANPA (ie 1NXX) but charged $2511/minute to callers. Because they are not in the US the FTC rules about declaring that it is a premium service number and the charges when first called do not apply. There are only a couple area codes 809 seems to come to mind but I cant guarantee that. In short you might investigate a phone company service blocker for premium service numbers and try your best to block what you can but it would be impossible for someone without SS7 network access to see what the rate of the call is since these numbers can hide virtually anywhere. I tried googling this topic, but it's hard to find anything with such common keywords. If anyone can direct me to a good resource, I'd appreciate it as well. NANPA manages all the numbers in the north american numbering plan, if memory serves their page is nanpa.org and they used to have rate center information available on their page for free that you can download (and you would need to parse it and continually get updates as new exchanges are allocated). On athe same topic, I'm worried about area codes like 809. Are there any other such area codes that should be avoided? Ahh glad you brought that up, see above. I think there are a couple of them, but I dont know off hand what they are.. try googling 'toll fraud 809' and see if that works. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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
