Re: [asterisk-users] OT: Free DID/SIP accounts
Roderick A. Anderson wrote: Actually the in the US. Inland Northwest. North Idaho if anyone is interested. http://www.ipkall.com - Free WA state DID numbers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dail in modem
Umm.. they call that a BBS. :-) Sounds like the perfect application for FidoNet/Opus... ABBAS SHAKEEL wrote: Sorry i replied late bcz i have to do some other work I have a new required functionality. that is Develop a Client server application that will communicate using a normal modem with out connecting to internet.(Client with a PC and modem will dail the number of server it will be a PSTN number (Not an ISP like thing) and the server with modem will recieve the call and receive some data and return results). Direct communication like hyper terminal. no connection to internet. i have tried TAPI(C#) and JTAPI (java) but dont get sucess. I am thinking Asterisk can handle that using TDM 400P card regards Shakeel Abbas On Sat, Jun 20, 2009 at 7:19 PM, Geraint Leegera...@gmail.com wrote: If i understand correctly you need users to be able to dial in using a modem to your servers then you are going to share your internet connection with those who dial your server. So, no, it has nothing to do with asterisk... you want to be looking at wvdial for the clients (assuming they are linux) and whatever the equivalent server would be (don't know as i've never done it). Good luck 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Geraint lee I also dont know .what kind of requirements are these :P i am just looking if it can happen On Fri, Jun 19, 2009 at 9:33 PM, Geraint Leegera...@gmail.com wrote: is it just me or am i right in thinking this has nothing to do with asterisk? 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Hello Actually i am required to make two application 1) that user use 2) that is deployed on server Application for user will be just like the windows standard connection using dail up modem but user will dail my PSTN number instead of the number we inter provided by ISP. on deployed server side we will get he usename and pass and other parameters of application and then use them in java code is it possible ? (nothing is impossible but for a Asterisk and java developer with limited time frame) Thanks On Fri, Jun 19, 2009 at 7:24 PM, Bob Piercepier...@westmancom.com wrote: On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISP -Asterisk - ATA -DIALUP
Not true... You can provided you disable data compression (ATK0) on your modem. Reason? Because a codec is already compressed. Adding compression at the modem level to an already compressed bitstream == lost bits. I call all over the world all the time using asterisk/sip/ulaw with decent bit rates. Alex Balashov wrote: Without getting into a lot of detail, this will not work. Period. You just can't do reliable modem passthrough with VoIP in most cases, some clever proprietary hacks notwithstanding. To the extent it is possible, nobody is going to send you the procedure.. This list is for specific answers to specific questions. -- Sent from mobile device On Jun 29, 2009, at 10:47 AM, Vidura Senadeera vidura...@gmail.com mailto:vidura...@gmail.com wrote: Hellow, / I have a problem with dial up signalling. currently I have configured asterisk server and E1 card to ISP. then other side I am having ATA to PC for connecting internet through DialUP connection. is it possible and please send me the procedure how I can do it ?? / ISP - Asterisk - ATA - DIALUP -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX based war dialer
It utilizes the iaxclient for piping the raw audio to a flat file where it's then analyzed. On Fri, Mar 6, 2009 at 11:43 AM, Alex Balashov abalas...@evaristesys.comwrote: Last thing we need is more war. -- Sent from mobile device On Mar 6, 2009, at 2:29 PM, Steve Edwards asterisk@sedwards.com wrote: This may be of interest -- as a tool we can use to test our systems and as a weapon that may be used against us :) http://warvox.org/ A brief read-over looks like it uses iaxclient and ruby to war dial a range of numbers and record audio samples to be analyzed to identify if the call was answered by a modem, fax machine, human, etc. The calls are placed through a PSTN termination provider. I didn't see anything about IAX brute forcing. SIP was mentioned, but the primary focus appears to be IAX providers. Thanks in advance, --- - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX based war dialer
Umm... Caller ID spoofing and DSP audio processing of called numbers are two entirely different subjects. And as far as creating more laws: I say fix the damn technology first (Caller ID) before wasting tax payers money on more laws on the books that will be obsolete in a few years time. While on the subject, it shouldn't be against the law to spoof one's own numbers. Other person's numbers, sure.. but if I want all my numbers to appear from one line, that's my own business. CID is sent in-band to a device *prior* to the ring voltage. And sometimes the data is sketchy depending on where your telco gets it's LIDB dips from. As for your last point, I'm sure you'd also like to see the death penalty for jaywalking because you had to tap your car brakes or be late by 5 seconds. Remind me not to waste my brake pads when you walk across the street. On Fri, Mar 6, 2009 at 1:24 PM, Jon Pounder j...@inline.net wrote: Tim Nelson wrote: The fact that this would be even being discussed on this list is an embarrassment to the asterisk community. I am constantly being pestered by cold callers with fake caller ids, probe calls such as this, etc. I think for once CRTC/FCC need to step up to the plate and take some useful measures : - make knowingly presenting forged caller id a federal crime (its fraud and harassment already) - block caller id spoofing at the telco boundaries (we all do this now for ip addresses, so why not caller id ?) - ban offenders from having telecommunications service of any sort nationally once convicted. If the telcos can't adapt to providing service and accountablity this way and actually serving the customers who pay them, telecommunications with just evolve without them. Much the way the post office is being left behind since they can not compete with the speed of fax and email for documents or couriers for packages. Another war dialer with IAX capabilities: http://www.softwink.com/iwar/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Steve Edwards asterisk@sedwards.com wrote: This may be of interest -- as a tool we can use to test our systems and as a weapon that may be used against us :) http://warvox.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX based war dialer
In Canada the do not call registry is useless since calls do not originate in Canada nor do the violators care if they are doing something illegal, Telcos could take this further and if a number of complaints are received about a call source, offer an opt-in blocking plan to throw those calls away, and simply answer them with a sorry you call is blocked since you have been blacklisted (just like known spam sources). Why do people have inherent trust on Caller ID? Why are entire systems built on the premise that Caller ID is legitimate data? The telco's developed Caller ID as a service to satiate the customer's demand of knowing who's calling before answering. The problem is that they don't pull from the same database (CID is *NOT* ANI). Nor do they work at the same level (CID == Inband delivery vs. ANI == Trunk Accounting). Telco's already provide a service for blocking numbers that aren't on a pre-approved list. But the better answer is to actually *enforce* the laws *already* on the books... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AS5200 - T100P - No alarms but no calls either...
No data is logged on the call. Probably because the status is reporting down *CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 *CLI pri debug span 1 Enabled debugging on span 1 *CLI Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended [] --- # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCT1/0 Digium Wildcard T100P T1/PRI Card 0 (MASTER) span=1,0,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 # Global data loadzone= us defaultzone = us --- [channels] language=en context=internal switchtype=national signalling=pri_net usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes callprogress=yes channel = 1-23 Tzafrir Cohen wrote: On Tue, Nov 11, 2008 at 06:02:49PM -0800, Don Fanning wrote: Greetings, I have a AS5200 that I have interfaced to a T100P via a T-1 Crossover cable. I got it where the alarms are all ok/green but I'm unable to dial out or dial into the AS5200. Anyone have any suggestions as to where to begin troubleshooting this? pri show span 1 pri debug span 1 And then see what happens on a call. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AS5200 - T100P - No alarms but no calls either...
Greetings, I have a AS5200 that I have interfaced to a T100P via a T-1 Crossover cable. I got it where the alarms are all ok/green but I'm unable to dial out or dial into the AS5200. Anyone have any suggestions as to where to begin troubleshooting this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual or Hardware SIP Modem
Short answer: No. However you can use ATA devices like PAP-2's to connect to your existing modem bank and as long as your latency is constant, get decent results. I myself have gotten 33.6k on a regular basis with such a setup and have called the world using cheap SIP/IAX providers with decent speeds. The key to note is that you disable Data Compression (ATK0) because the data stream is already compressed. Error correction however is useful. Kyle Gibbons wrote: Hi, I have just gotten my first Asterisk box up and running, and it is running great. I am working on this project with the plans of possibly implementing it in a business environment. The problem I am coming up against is that the business I am planning on implementing this setup in is using some legacy software which requires a modem to communicate with energy management systems. My question is if there is a virtual or physical SIP modem that I could possibly use so that I can interface this old software with Asterisk. There is no option of getting rid of modems all together. I would prefer not to use Zap cards or other adapters for the current modems. My goal is to completly replace the modems with software. Any help would be GREATLY appreciated. Thanks! -- All the best, Kyle bobert5064.deviantart.com http://bobert5064.deviantart.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisc for Taking Calls for Radio
Umm... create your dial plan then use a softphone? You'll have to work out the audio connections in and out of your computer that feeds a audio channel and outputs the monitor back to the computer but it can be done pretty easily. Shane D wrote: Hello Asterisc-Users List, I am new to the list. I joined with a question in mind: How would you set up an asterisc box so that: (A) Someone dials a number (B) They are presented with a menu (C) Entering a number, like 1, connects a call to me. (D) I am on a mixing board, running an internet radio show. I want to run asterisc into the board, and run an output from the board to asterisc. Is that possible using a soundcard? I don't really want to spend money. (E) I want the board to start wringing when I get a call, and I want the call audio to the board as well. I also would like it if I could not use my local phone line. I would prefer something like a free internet based number. The box will not need to be able to call out, so that's not a problem. A friend of mine uses asterisc, and has a free internet based number for asterisc. I would like to do the same. I hope this is possible, and thanks in advance. Shane ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?
Horse hockey... I currently have a *BANK* of PAP2's hooked up to a wide array of analog modems (a USR Total Connect MP/8, two USR Courier V.Everythings and a Digi LANASERVER). After balancing the audio on the pap2's to not feedback audio and reduce chances of echo occurring, I've had no problem maintaining all lines running whether it's within the LAN or from West Coast USA to Europe (fidonet bbs's and x.25 networks) or between the West Coast USA and Australia via SIP point-to-point. The max speed i've obtained is 33.6kbits/s and that's the normal maximum for *non-ISP* configurations. The key things to setup for is: 1.) Steady latency. Latency is the line killer because modems rely on timing. Most of the time (95%) it's not an issue as my routes to the various VSP's I use have a constant strain/timing between myself and them. 2.) Disable Data Compression on the modem and save it in the NVRAM of the modem. (ATK0) Digitized analog signal already has enough lost bits. *DO* however leave Error Correction on. If both modems support it, it helps tremendously even through lag events. 3.) Test, test and retest... Listen to the connection. If it doesn't work at faster speeds, use the ATNx where x is a number from 0 (auto) to 1 (300bps) to 2 (1200bps) etc... so you can figure out the maximum potential of your hardware and voip connections. So yes Virginia, you can do analog modems over VoIP without issue. And pull a decent data rate. All you would need then is to configure the modem and the machine it's connected to as a PPP server then configure the phone to call your modem via *. Anselm Martin Hoffmeister wrote: Am Dienstag, den 27.11.2007, 18:00 +0100 schrieb Robert Rozman: Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come to Asterisk and I'd love to somehow fool that device and connect it to local web pages ? I guess I could somehow mimic ISP internet calling feature on local Asterisk server, but have no clue even where to start searching ... Any advice ? Hi Robert, I researched for something similar about a year ago, and came up with nothing really worth the work. If you can, try to get another ATA that has a real, old-fashioned serial modem plugged into it, and limit that modem to 9600. I think more than that will not work reliably, but you could of course try. The only working implementation of software emulating a modem in conjunction with asterisk I have seen is fax-related, and even there I read from several people that anything better than 9600 is hardly ever achieved. The code there is cranked into fax-use though, not modem use, which would require the PPP bytestream to be off-handed instead of fax parsing. Perhaps iaxmodem would do that No idea. I'd be interested in how you get that working, if you do indeed. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USB Modem with asterisk
No. A USB modem will *not* work as a FXO card. You're thinking of the X100P card which was a rebranded Intel modem. These have been discontinued by Digium and only third party suppliers still sell them. But taking a normal modem and using it as a FXO will not work (most modems do not pass audio information to the system bus). Additionally, having multiple modems won't help you either. You'll need multiple cards with multiple FXO ports on them. A better solution in this case would be to get a TDM card and run it into your T-1 CSU/DSU (or router). Doug Zingel wrote: I can use a USB modem with asterisk to connect to the PSTN network right? It'll serve the same functionality as an FXO card? Also, any idea if I can get these modems with mutiple ports (12 or 24)? Thanks, Doug Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Don Fanning http://00100100.net Email Fortune: It may or may not be worthwhile, but it still has to be done. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Do Not Call List
This part below did say you can be a 3rd party. If the telemarketer is accessing the registry on behalf of other sellers or telemarketers, that telemarketer also must identify each of the other sellers or telemarketers on whose behalf it is accessing the registry, and it must certify, under penalty of law, that the other sellers or telemarketers will be using the information gathered from the registry solely to comply with the provisions of this rule. -Original Message- From: Kevin Bockman [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Sunday, November 19, 2006 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Do Not Call List Keep reading. The person that actually does the calling needs to be registered. You can't provide the list to others either. Kevin Don Fanning wrote: Oddly enough, there's really nothing stopping one from doing so in the material I just scan through at: http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm In regards to the fee, here is the latest: The amended rule increases the annual fee for access to the Registry for each area code of data to $62 per area code, or $31 per area code of data during the second six months of an entity's annual subscription period. The maximum amount that would be charged to any single entity for accessing 280 area codes of data or more is increased to $17,050. In addition, the amended rule retains the provisions regarding free access by exempt organizations, as well as free access to the first five area codes of data by all entities. In particular, here is the part on the usage... If a central database (external from the FTC) does start up, they'll have to register who uses the database. --- § 310.9 Fee for access to do-not-call registry. (c) Access to the do-not-call registry is limited to telemarketers working on their own behalf or working on behalf of other sellers or telemarketers. Prior to accessing the do-not-call registry, a telemarketer must provide the identifying information required by the operator of the registry to collect the user fee, and must certify, under penalty of law, that the telemarketer is accessing the registry solely to comply with the provisions of this rule. If the telemarketer is accessing the registry on behalf of other sellers or telemarketers, that telemarketer also must identify each of the other sellers or telemarketers on whose behalf it is accessing the registry, and it must certify, under penalty of law, that the other sellers or telemarketers will be using the information gathered from the registry solely to comply with the provisions of this rule. -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Friday, November 17, 2006 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List I'm surprised someone doesn't come up with a consortium for all the asterisk users to poll a central location or does the data come with restrictions about sharing the data? Duane from e164.org says he's already built the application you are looking for to deal with Australian databases if that helps. Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users _ ella for Spam Control has removed 3979 Spam messages and set aside 119 Newsletters for me You can use it too - and it's FREE! www.ellaforspam.com http://www.ellaforspam.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Do Not Call List
I have a request into their operations @ the FTC asking for developer access to write a module based on their data. We'll see... -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Sunday, November 19, 2006 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List Thanks for looking into this further Kevin. I guess this knocks a 'formal' asterisk asp sharing agreement on the head. I can understand why they have done this but also sucks for people installing asterisk using this. At least the formal data sets are documented so a module for lookup prior to calling can be checked against. I haven't checked as this isn't my space but I guess anyone offering predictive dialers to asterisk is already building this into their product offerings (or coding as we speak). Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Bockman Sent: Sunday, 19 November 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Do Not Call List Keep reading. The person that actually does the calling needs to be registered. You can't provide the list to others either. Kevin Don Fanning wrote: Oddly enough, there's really nothing stopping one from doing so in the material I just scan through at: http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm In regards to the fee, here is the latest: The amended rule increases the annual fee for access to the Registry for each area code of data to $62 per area code, or $31 per area code of data during the second six months of an entity's annual subscription period. The maximum amount that would be charged to any single entity for accessing 280 area codes of data or more is increased to $17,050. In addition, the amended rule retains the provisions regarding free access by exempt organizations, as well as free access to the first five area codes of data by all entities. In particular, here is the part on the usage... If a central database (external from the FTC) does start up, they'll have to register who uses the database. --- § 310.9 Fee for access to do-not-call registry. (c) Access to the do-not-call registry is limited to telemarketers working on their own behalf or working on behalf of other sellers or telemarketers. Prior to accessing the do-not-call registry, a telemarketer must provide the identifying information required by the operator of the registry to collect the user fee, and must certify, under penalty of law, that the telemarketer is accessing the registry solely to comply with the provisions of this rule. If the telemarketer is accessing the registry on behalf of other sellers or telemarketers, that telemarketer also must identify each of the other sellers or telemarketers on whose behalf it is accessing the registry, and it must certify, under penalty of law, that the other sellers or telemarketers will be using the information gathered from the registry solely to comply with the provisions of this rule. -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Friday, November 17, 2006 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List I'm surprised someone doesn't come up with a consortium for all the asterisk users to poll a central location or does the data come with restrictions about sharing the data? Duane from e164.org says he's already built the application you are looking for to deal with Australian databases if that helps. Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users _ ella for Spam Control has removed 3979 Spam messages and set aside 119 Newsletters for me You can use it too - and it's FREE! www.ellaforspam.com http
RE: [asterisk-users] Do Not Call List
A quick google search says there isn't anything written yet. But looking at the database itself, it seems pretty easy to import data into a sql table or do xml pulls from them directly.. https://www.donotcall.gov/FAQ/FAQBusiness.aspx#download https://www.donotcall.gov/FAQ/FAQBusiness.aspx#download It wouldn't be hard to code up at all actually... a little perl magic and voila. ;) Who needs a weekend project? -Original Message- From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Wednesday, November 15, 2006 9:18 PM To: Asterisk-Users Subject: [asterisk-users] Do Not Call List The US has a Do Not Call list to which people can subscribe to prevent being called by advertisers. Federal laws (strengthened by some state and more local laws) assign penalties for calling people/phones on the DNCL. Is there a query gateway that Asterisk (or an app using Asterisk) can filter through to ensure a number is OK to call (not on the list) before calling it? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users _ ella for Spam Control has removed 3785 Spam messages and set aside 117 Newsletters for me You can use it too - and it's FREE! www.ellaforspam.com http://www.ellaforspam.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Do Not Call List
Depending on your organization, you're allowed up to 5 area codes for free. -Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Friday, November 17, 2006 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List It wouldn't be hard to code up at all actually... a little perl magic and voila. ;) Who needs a weekend project? The Perl magic would be easy. Writing the check to pay for all of that data is what is so hard... -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users _ ella for Spam Control has removed 3789 Spam messages and set aside 117 Newsletters for me You can use it too - and it's FREE! www.ellaforspam.com http://www.ellaforspam.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Do Not Call List
Oddly enough, there's really nothing stopping one from doing so in the material I just scan through at: http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm In regards to the fee, here is the latest: The amended rule increases the annual fee for access to the Registry for each area code of data to $62 per area code, or $31 per area code of data during the second six months of an entity's annual subscription period. The maximum amount that would be charged to any single entity for accessing 280 area codes of data or more is increased to $17,050. In addition, the amended rule retains the provisions regarding free access by exempt organizations, as well as free access to the first five area codes of data by all entities. In particular, here is the part on the usage... If a central database (external from the FTC) does start up, they'll have to register who uses the database. --- § 310.9 Fee for access to do-not-call registry. (c) Access to the do-not-call registry is limited to telemarketers working on their own behalf or working on behalf of other sellers or telemarketers. Prior to accessing the do-not-call registry, a telemarketer must provide the identifying information required by the operator of the registry to collect the user fee, and must certify, under penalty of law, that the telemarketer is accessing the registry solely to comply with the provisions of this rule. If the telemarketer is accessing the registry on behalf of other sellers or telemarketers, that telemarketer also must identify each of the other sellers or telemarketers on whose behalf it is accessing the registry, and it must certify, under penalty of law, that the other sellers or telemarketers will be using the information gathered from the registry solely to comply with the provisions of this rule. -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Friday, November 17, 2006 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List I'm surprised someone doesn't come up with a consortium for all the asterisk users to poll a central location or does the data come with restrictions about sharing the data? Duane from e164.org says he's already built the application you are looking for to deal with Australian databases if that helps. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Friday, 17 November 2006 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List It wouldn't be hard to code up at all actually... a little perl magic and voila. ;) Who needs a weekend project? The Perl magic would be easy. Writing the check to pay for all of that data is what is so hard... -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users _ ella for Spam Control has removed 3790 Spam messages and set aside 117 Newsletters for me You can use it too - and it's FREE! www.ellaforspam.com http://www.ellaforspam.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digium GUI?
You mean the menuselect ncurses screen? If yes, then yes... it's a gui. :) -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Monday, September 18, 2006 4:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Digium GUI? So the press announcement said that the new Digium GUI will be available in v1.4 sometime in Oct. Is the GUI already there in Trunk or is there some other branch of development that the general public cannot access? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Connecting an cellphone to asterisk
How about a Cell Socket? Just plug it into your FXO card and you're set. http://www.ctdi.com/cellsockets.htm - Original Message - From: Alvaro Cornejo [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, August 20, 2006 1:38 PM Subject: [asterisk-users] Connecting an cellphone to asterisk Hi Is there a way to connect an Cellphone to asterisk in order to route calls though it?. This is what I want to do: Here is much cheaper to call from cell to cell than from fixed line to cell. So I want to connect a cell to the asterisk box and create a rule to route calls to a cell through the cell connected to the asterisk box. Is it possible? Can I do it with the standard data USB cell-pc or I need a special cable/connection? Did someone worked this? Wich cell brand/model can I use for that? Any tips would be appreciate. Regards Alvaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
Use a virtual private asterisk system. You'll be happier if you did. http://www.telephreak.org/papers/vpa/ Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. It's obvious that Asterisk was designed more for the enterprise (ie a single company), rather than for the carrier (ie multiple companies). It's a bit hard to explain here, but even with more than a few companies, the config files and dial plan start to become horribly complex. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
The manager interface isn't some mystical beast that can't be overcome. Try the wiki if you're lost. Really people scripting isn't that hard. If you don't like the way people do code, there's nothing stopping you from writing something new (except for lack of skill but that's why people do it for a living). You buy them books, send them to school and all they do is eat the pages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, August 16, 2006 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [asterisk-users] Manager Interface API's Actually, because there's no documentation, I don't have anything that I can use. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 12:54 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Some of them write it for them selves and out of the goodness of thier heart will put out there for free. They dont need doc's since they wrote it them selves. Be happy that you got it for free. Do you want people to stop releasing code because others complain ? - Original Message - From: John Novack [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 15, 2006 12:39 PM Subject: Re: [asterisk-users] Manager Interface API's I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough information for many others to make effective use of their creation How many have struggled through the years with uncommented or poorly commented code when the original creator is off to greener pastures? JMO John Novack Moises Silva wrote: Douglas. Please take this as a constructive comment. I have followed your questions in asterisk-dev and users lists, and you always seem to make non constructive comments about the people giving code/work for Free. And you focus in the negative part, never giving importance to the positive things about it. If you dont like something, then change it yourself, they are not providing a payed service. The source is available AS-IS if you want it, and if you like it, take it; If you dont, just ignore it, try to not make peyorative comments. Regards On 8/15/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, I don't know about you, but if I have to read the source code to work out how it works, I'm going to go and look at someone elses, that may have some BASIC documentation and examples. -Original Message- From: Don [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Probably cause it is someone like most of us sitting at home doing it...releasing it for free...so why would we write pages of documentation for it? If it's open source and it's free...Then offer them some money to make documentation for it hehe... - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 15, 2006 11:05 AM Subject: [asterisk-users] Manager Interface API's Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: 8/15/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
[asterisk-users] Ringing after answered on zaptel
Greetings List, Im having a strange problem with my X100p card still ringing after the call is connected. Any idea on how to solve this? Using latest asterisk (not svn) along with latest zaptel driver. Thanks, Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterfax and Gentoo
Greetings List, Anyone got this working with Gentoo? Or at least a howto to run it on systems NOT running trixbox? Thanks, Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterfax and Gentoo
Alright, Cool. Ive followed the directions but the java application is erroring out with some funky issues relating to directories/permissions/broken components. --- 14 Aug 2006 21:57:01,821 DEBUG FaxManagerOutBound passed message to channel Zap/g0 Both from-internal priority(1) count(1) for delivery. 14 Aug 2006 21:57:01,842 ERROR java.lang.NullPointerException at au.com.noojee.asterfax.outbound.Fax.getFrom(Fax.java:317) at au.com.noojee.asterfax.outbound.Fax.toString(Fax.java:411) at au.com.noojee.asterfax.outbound.Channel.run(Channel.java:95) 14 Aug 2006 21:57:01,897 ERROR Error attempting to send error response java.lang.NullPointerException at au.com.noojee.asterfax.Responses.toString(Responses.java:344) at au.com.noojee.asterfax.Responses.sendError(Responses.java:205) at au.com.noojee.asterfax.Responses.sendError(Responses.java:121) at au.com.noojee.asterfax.outbound.Channel.run(Channel.java:129) 14 Aug 2006 21:57:01,899 DEBUG Preserving spool file: fax 14 Aug 2006 21:57:01,908 ERROR Error attempting to preserve spool file java.io.FileNotFoundException: /var/spool/asterfax/tmp/fax (No such file or directory) at java.io.FileOutputStream.open(Native Method) at java.io.FileOutputStream.init(Unknown Source) at java.io.FileOutputStream.init(Unknown Source) at au.com.noojee.asterfax.util.FileSystem.copyFile(FileSystem.java:75) at au.com.noojee.asterfax.util.FileSystem.moveFile(FileSystem.java:108) at au.com.noojee.asterfax.messagestore.FileMimeMessage.moveFile(FileMimeMessage.java:306) at au.com.noojee.asterfax.outbound.Fax.moveToTemp(Fax.java:192) at au.com.noojee.asterfax.outbound.Channel.run(Channel.java:151) 14 Aug 2006 21:57:01,909 DEBUG Making channel Zap/g0 Both from-internal priority(1) count(1) available. 14 Aug 2006 21:57:03,181 ERROR Authentication failed org.asteriskjava.manager.AuthenticationFailedException: Authentication failed at org.asteriskjava.manager.DefaultManagerConnection.login(DefaultManagerConnection.java:471) at org.asteriskjava.manager.DefaultManagerConnection.login(DefaultManagerConnection.java:365) at au.com.noojee.asterfax.inbound.FaxManagerInbound.connect(FaxManagerInbound.java:166) at au.com.noojee.asterfax.inbound.FaxManagerInbound.init(FaxManagerInbound.java:85) at au.com.noojee.asterfax.inbound.FaxManagerInbound.getInstance(FaxManagerInbound.java:71) at au.com.noojee.asterfax.AsterFax.startThreads(AsterFax.java:190) at au.com.noojee.asterfax.AsterFax.init(AsterFax.java:114) at au.com.noojee.asterfax.AsterFax.main(AsterFax.java:297) 14 Aug 2006 21:57:03,182 INFO Shutting down AsterFax due to unexpected exception. Check logs/AsterFax.log for details. 14 Aug 2006 21:57:03,183 INFO Outbound FaxManager Stopping 14 Aug 2006 21:57:03,184 INFO AsterFax has shutdown. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warrick Zedi Sent: Monday, August 14, 2006 9:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterfax and Gentoo There are instructions to install from source packages at http://asterfax.sourceforge.net/Installing%20AsterFax.html in the Manual section. Those instructions do refer to installing trixbox and using yum but you can ignore the trixbox instruction and use your favourite package manager. If you cant locate appropriate versions of packages such as Java, ghostscript and openoffice.org using your package manager then manually locate them, download and install them. When you get to the AsterFax install get the zip file (which I only just made available for rc5) and unzip the file to /usr/lib/asterfax then edit config/AsterFax.xml and bin/ooconvert.sh checking that the paths to ghostscript and openoffice are correct for you install. Most of this is relatively straightforward. The only tricky part is getting spandsp patched.If you use spandsp0.0.2pre26 then AsterFax comes with an already patched app_txfax.c that you can just copy over the existing file and that should work. Let me know how you go. We are planning on providing debs and Ill add gentoo to that list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Fanning Sent: Tuesday, 15 August 2006 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterfax and Gentoo Greetings List, Anyone got this working with Gentoo? Or at least a howto to run it on systems NOT running trixbox? Thanks, Don -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Q: How to dial out / transfer calls with manager
Greetings, Here's my issue. My local free VSP isn't transfering proper DTMF (inband or converting to RFC2833) so I'm stuck with making a php interface so my roommates whom are not using softphone/ata devices to call out via * (and thusly get the better deals in Long Distance). I've tried using the Manager interface to creating the connection however when I create a Channel: it needs to be something virtually attached to the system. I'm trying to see if there is a way around it. IE: Currently I drop fputs($socket, Secret: ibanez\r\n\r\n); fputs($socket, Action: Originate\r\n); fputs($socket, Channel: $mytelephone\r\n); fputs($socket, Exten: 1$callnumber\r\n); fputs($socket, Priority: 1\r\n\r\n); From a php script with $mytelephone being the home phone via sip like SIP/[EMAIL PROTECTED] and $callnumber is the destination number which would default to my $TRUNK. However since the channel isn't registered on the system it will fail. Is there a way of cheating this via callpark or meetme? How about a dummy iaxclient to originate then dumps to a meetme with the $callnumber doing the same? I find this very limiting as I can't route calls the way I want to. (the DTMF issue is worse... Don't get me started. ;) Ideas? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Q: How to dial out / transfer calls with manager
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Monday, January 02, 2006 7:26 AM Your problem is related to not reading the documentation in voip-info.org :) Umm.. Yeah I have. Otherwise I wouldn't be a pain in the ass right now. I'd just be clueless. :) You can originate a call to anyplace doing: - First a LoginAction. - Then an Originate action with the proper arguments. In the example you put, you are doing neither of them. You can test manually how the protocol works doing a login from a telnet client: telnet localhost 5038 (in case your in the asterisk box) Action: Login\r\n Username: someuser\r\n Secret: somesecretpassword\r\n\r\n I can see myself login to the manager port just fine (even after I changed my password from my post slippage ;) Action: Originate Channel: SIP/13 -- this should be the first phone you want to ring (your own phone usually) I don't want it to ring a REGISTERED device (SIP/IAX/ZAP) that is on the system. I want it to make a outbound call externally through my VSP and when it's answered, then make another outbound call on another channel. Context: somecontextwithoutbountpatterns Not essential Exten: --- extension that will make your call Priority: 1 (usually one is fine) Again, Ideas? Thanks, Don ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I thought they weren't charging - FW: [DIDx.net] Happy holidays wishes from DIDX.net.
Did anyone else get this? I thought they weren't charging? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 28, 2005 9:35 PM To: x Subject: [DIDx.net] Happy holidays wishes from DIDX.net. Dear x, The DIDX.net team wishes you a very happy holiday season. DIDX has revised its monthly rates' structure. We will no longer charge you anything to be a Regular Member in the DIDX network. Once you are comfortable with DIDX and are ready to start your trading on the DIDxchange, you will be required to keep a minimum of 20 DID's buy or sell total. This is a Regular Membership. Otherwise, you will be charged a minimum monthly fee of $20. You can avoid this charge by purchasing 20 DID's for as low as 10 cents each. This will total $22 a month for 20 DID's with our commission charges. Thank you for joining and being a part of the successful DIDX network, the fastest growing VOIP exchange in the world. * To un-subscribe to our news letter, Please login to your account, click on edit my info, and you can unscubscribe to this news letter from there. You can not un-subscribe to our notification emails.. RefFile: DIDx - Email.pm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDD/TTY - How does one use this?
I'm trying to look for documentation on how the TDD/TTY interfaces with the user. From the looks of it, fskmodem talks directly to a channel. Does it matter what type of channel it connects to? SIP/IAX/Zap? Secondly, how does one interface with it on the asterisk side? Obviously there is no sendtty function in the cli and it would be the wrong place for it. How does it work? Thanks -Don ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to receive DTMF inbound
Greetings, I'm trying to get * to work on my inbound SIP connections through callpacket. Currently I'm getting the following issue when I hit a DTMF on the inbound channel. -- Playing 'agent-pass' (language 'en') Dec 9 14:07:42 NOTICE[13642]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 67.43.155.130 -- Playing 'auth-incorrect' (language 'en') When I have it call my softphone and play dtmfs from the inbound it sounds like a buzz rather than a tone like it's overmodulated and then scatters. Of course I get the standard line from callpacket (sorry - can't help you) so how can I resolve this? I've tried applying the two bug fixes listed in the digium lists but no effect. Ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF is choppy on the receive
I'm currently using X-Ten as a softphone and I've been having issues with dialing into IVR's. It seems that my DTMF passes in chirps and not clear tones. Any solutions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipbuster
I ended up buying a second 1 euro account because of this. But it does work fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Thursday, December 01, 2005 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voipbuster I was testing voipbuster. With a new account, with no credit, I can make calls perfectly but of 1 minute. But I tried the username and passwrord of an account with credit, and the registration is refused. With the voipbuster propietary software it works ok (I sniffed the packets and I think it is not using standard iax or sip ports). Are the acconts with credit blocked for avoiding it's use with ohter software than voipbuster's? I tryed to send a mail to voipbuster's support but I never received an answer (then do not support other thing than their software). -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Virtual Modems Revisited
I brought this up a while back and althought there are pieces that interface * into Fax Telephony applications, there hasn't been something that works with plain old analog modems. Then I found this piece of code. From my initial tests it looks solid, but I have no clue in how to interface this into asterisk. I thought I would put this link up for other people to comment and try. http://fabrice.bellard.free.fr/linmodem.html Out of the box it works with soundcards. I've been battling jack and alsa for a week trying to get them to play nice just to reroute the audio but I'm out of time in this regard. So I thought I would toss it up and see what other people can come up with. Happy Holidays! Don ___ --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] Virtual Modems Revisited
Whoops... Sorry.. Mailer delay. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Fanning Sent: Wednesday, November 23, 2005 5:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Virtual Modems Revisited I brought this up a while back and althought there are pieces that interface * into Fax Telephony applications, there hasn't been something that works with plain old analog modems. Then I found this piece of code. From my initial tests it looks solid, but I have no clue in how to interface this into asterisk. I thought I would put this link up for other people to comment and try. http://fabrice.bellard.free.fr/linmodem.html Out of the box it works with soundcards. I've been battling jack and alsa for a week trying to get them to play nice just to reroute the audio but I'm out of time in this regard. So I thought I would toss it up and see what other people can come up with. Happy Holidays! Don ___ --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
RE: [Asterisk-Users] Virtual Modems Revisited
Ah... Well I was sort of thinking more along the lines of trying to get this to work into IAX or SIP. But if you know for sure that the modulation is broken... Just imagine... You'd be able to have a modem bank and save thousands of dollars in leasing/purchasing a modem bank. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Wednesday, November 23, 2005 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Virtual Modems Revisited Don Fanning wrote: I brought this up a while back and althought there are pieces that interface * into Fax Telephony applications, there hasn't been something that works with plain old analog modems. Then I found this piece of code. From my initial tests it looks solid, but I have no clue in how to interface this into asterisk. I thought I would put this link up for other people to comment and try. http://fabrice.bellard.free.fr/linmodem.html Out of the box it works with soundcards. I've been battling jack and alsa for a week trying to get them to play nice just to reroute the audio but I'm out of time in this regard. So I thought I would toss it up and see what other people can come up with. Happy Holidays! Don Linmodem doesn't work out the box with anything. linmodem was abandoned by its author at a very early stage, before any of its component parts really worked. It has a number of useful bits, which might be used as the basis for a modem. It does not have a properly working code for any of the modem standards. I think Fabrice got busy, and with patent issues preventing wide deployment of a V.34 modem finished the software just seemed like a waste of time to him. He is one of the good guys of free DSP, and has since produced several valuable things which are complete. Steve ___ --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
[Asterisk-Users] Virtual Modems Revisited
I brought this up a while back and althought there are pieces that interface * into Fax Telephony applications, there hasn't been something that works with plain old analog modems. Then I found this piece of code. From my initial tests it looks solid, but I have no clue in how to interface this into asterisk. I thought I would put this link up for other people to comment and try. http://fabrice.bellard.free.fr/linmodem.html Out of the box it works with soundcards. I've been battling jack and alsa for a week trying to get them to play nice just to reroute the audio but I'm out of time in this regard. So I thought I would toss it up and see what other people can come up with. Happy Holidays! Don ___ --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] Virtual Modems Revisited
I brought this up a while back and althought there are pieces that interface * into Fax Telephony applications, there hasn't been something that works with plain old analog modems. Then I found this piece of code. From my initial tests it looks solid, but I have no clue in how to interface this into asterisk. I thought I would put this link up for other people to comment and try. http://fabrice.bellard.free.fr/linmodem.html Out of the box it works with soundcards. I've been battling jack and alsa for a week trying to get them to play nice just to reroute the audio but I'm out of time in this regard. So I thought I would toss it up and see what other people can come up with. Happy Holidays! Don ___ --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] voipbuster advise
Greetings, 1.) Voipbuster does not support T.38. If you can get a clean connect using G.711u then the answer is maybe. Latency will ice a analog connection. 2.) That's built into the dialplan at VoipBuster. It's doubtful they'll remove the routing charge message, but you could always ask their customer service. :-) -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Monday, September 26, 2005 3:27 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] voipbuster advise Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when I'm with clients. Any links, suggestions? Thanks -- .:FaberK:. ___ --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
RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM
Time and time again, emergency action drills take place in cities to target where their weaknesses are in "crisis" handling. Usually they involve planes crashing or explosions (mock of course). Obviously they were never prepared for this sort of disaster in their recovery plan. I've participated in a few ARES/RACES drills and have to say that much could be done to improve upon the "HAM" infrastructure. Most of the time, communications is coordinated through 1 repeater system. When this repeater goes down, of course people would switch comms to another but in a case like this, where all the repeater systems go down except for maybe one, there needs to be a better plan. In Amateur Satellite Service, these orbiting "Repeaters" employ a system called RUDAK where a chunk of spectrum is repeated. Obviously this isn't feasible in terrestrial repeaters but they dohave the ability to turn off radios and switch bands at will depending on operating conditions. With software controlled radio and Asterisk, the repeater system could be made to be more resilient to disaster by linking to other repeater systems via radio where it could connect outward. If you figure the overhead of a repeater's transmitter and receiver plus the controller, replaceing the controller with an asterisk based unit (integration) would make more sense as it would give the repeater system much more capabilities in the same footprint and power. Additionally, these repeater systems are located on hilltops with other radio systems so they should have emergency power available (if you've ever been to a hilltop repeater site, you'll know what I mean). I think the biggest thing that hurts ham radio's ability to react to a crisis is the lack of equipment and operators. Most of the traffic we pass is "Health and Welfare" with "Logistics" being the second to it. What defeats this is that in a disaster where local/high band long haul capabilities are diminished, is simply the one repeater that is functional because everything is squeezed onto one VHF/UHF repeater. Where I could see thing being improved? Installation of 802.11b/g WLAN under Part 97. It would allow for more users into the system, there are less hardware and power components and allows the system to be dynamically configured. Asterisk could play a huge role then as it's made for IP based traffic and could re-route in a split second. -Don From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael D SchelinSent: Saturday, September 10, 2005 10:20 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM The two best forms of communications in a real disaster and one always has been is #1 Ham radio. and #2 satellite telephone. Ham radio is global and has proven time and time again to be the most reliable when the infrastructer has been damaged. The U.S government is the biggest user of satellite telephones which is also becoming a valuable tool again when the communications infrastructure is down. It would be nice If Asterisk could be used but in this case but it's useless. People are displaced and most of the communications infrastructure for the city is unusable. I don't mean all of the telco's systems. It's the flood that wiped out most home and business systems. For us, The best thing that a provider can do is to have redundant servers in different cities. This should remind us all how fragile our lives are. Chris Travers wrote: Mark Phillips wrote: Hold on here folks, I'm guessing that the original poster of this thread isn't a member of his local RAyNet team. Whilst I don't profess to be an expert at this I have been doing emergency radio for quite some time and have seen service at the Lockerbie bombing, Docklands bomb, Ground Zero (I'm sure I'm a terrorist target y'know - they seem to follow me everywhere) and soon I'll be in Louisiana. In all of these events the KISS principle must and does prevail. We need a system that is a simple and energy efficient as possible. Building a network of * servers and Wi-Fi links is all very well but how are you going to power them? These are excellent points. I have a few interesting suggestions here The first is that the only obstacle to any sort of longer-range point to point line is merely power. This is true whether you are talking HAM or fiberoptics. Note that if you have the power, it would take disruption of the physical line to disrupt a fiber line. Note that DirectNIC in New Orleans remained operational without *any* downtime or loss of connectivity with the rest of the world. The suggestion that I have is for various areas to have dedicated civil emergency com units with strategic reserves of fuel (3-4 weeks worth), battery backups, etc. These units would have links (fiber, microwave, and/or satellite, better to
RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM
I can understand that. I'm a KL7 call so comms could mean the matter of someone getting picked up or freezing to death. It troubles me that radio site owners (the ones who hold the pink slip on the tower and hilltop) are not providing power. In AK, most of these sites are multihomed with fed, state and local radio systems so money is provided to maintain backup power. That being said, in that given area, maybe taking a cue from the Emergency Call boxes along the I-5 and I-15 and use solar panels to charge a battery backup system. That plus some power-stingy equipment could maintain a reliable radio network. Knowing that all of us on the west coast are just || close to the big one when sites like this loose power to the cellular equipment, guess who's still going to be operating? :) (not that they would be working well anyways since lines jam up) Anyways. A resiliant recovery plan that has been practiced and works will trump a "all-hands" effort anyday. -Don From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael D SchelinSent: Sunday, September 11, 2005 2:46 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM Don, I agree with you on many fronts. I come from a radio background and here in southern cal unless we fall into the sea nothing will take out all of the communications here including ham because we are not in low lying flat land and were too diversified, over 150 miles and as many mountain top sites. BUT,let me tell you about how bad the southern CA. radio site owners are becoming. We had a 4 day outage at a very large site where one of my radios is located. None of them care anymore about backup power. This happened this past week. We took up our own Generator because the site owner (a national site company) won't maintain an old one. My friend (a microwave isp ) fixed the site owners by adding oil and a new battery. That will take us out!Don Fanning wrote: Time and time again, emergency action drills take place in cities to target where their weaknesses are in "crisis" handling. Usually they involve planes crashing or explosions (mock of course). Obviously they were never prepared for this sort of disaster in their recovery plan. I've participated in a few ARES/RACES drills and have to say that much could be done to improve upon the "HAM" infrastructure. Most of the time, communications is coordinated through 1 repeater system. When this repeater goes down, of course people would switch comms to another but in a case like this, where all the repeater systems go down except for maybe one, there needs to be a better plan. In Amateur Satellite Service, these orbiting "Repeaters" employ a system called RUDAK where a chunk of spectrum is repeated. Obviously this isn't feasible in terrestrial repeaters but they dohave the ability to turn off radios and switch bands at will depending on operating conditions. With software controlled radio and Asterisk, the repeater system could be made to be more resilient to disaster by linking to other repeater systems via radio where it could connect outward. If you figure the overhead of a repeater's transmitter and receiver plus the controller, replaceing the controller with an asterisk based unit (integration) would make more sense as it would give the repeater system much more capabilities in the same footprint and power. Additionally, these repeater systems are located on hilltops with other radio systems so they should have emergency power available (if you've ever been to a hilltop repeater site, you'll know what I mean). I think the biggest thing that hurts ham radio's ability to react to a crisis is the lack of equipment and operators. Most of the traffic we pass is "Health and Welfare" with "Logistics" being the second to it. What defeats this is that in a disaster where local/high band long haul capabilities are diminished, is simply the one repeater that is functional because everything is squeezed onto one VHF/UHF repeater. Where I could see thing being improved? Installation of 802.11b/g WLAN under Part 97. It would allow for more users into the system, there are less hardware and power components and allows the system to be dynamically configured. Asterisk could play a huge role then as it's made for IP based traffic and could re-route in a split second. -Don From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael D SchelinSent: Saturday, September 10, 2005 10:20 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAMThe two best forms of communications in a real disaster and one always has been is #1 Ham radio
RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM
Try contacting the repeater trustee for http://www.wa3key.com/blura.html. They have a picture of one on their site with it lit up. You will need to recrystal the radio to a proper TX/RX pair for 70cm. However, depending on your area, you should contact your local repeater coordnator so you don't step on anyone's toes (especially the case in So.Cal ;) Looks like you can order crystals from: http://www.icmfg.com/motorola.html. And there are plenty of links associated with this hardware. Google is your friend. As for interfacing it to *, you'll need a phone patch adapter. You could purchase one or build one but you'll need to get more information on how to do such. Once you have the repeater up and running, you also need to setup * to see the phone patch/radio interface as a radio. This may require a controller card. (see the voip-info.org wiki) And... if you're going to go that far, consider enrolling into the echoirlp project. It's a VoIP oriented repeater link system that uses the internet as it's conduit. By Part 97 rule, the system must be protected from unlicensed use so interfacing with asterisk would require password protection and you as the repeater owner would be liable for any misuse of the system. 73 de Don From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve TotaroSent: Sunday, September 11, 2005 6:07 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM Just a shot in the dark here. I bought this unit http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5792377951rd=1sspagename=STRK%3AMEWN%3AITrd=1a couple months ago hoping to connect it to an * system for experimentation. I am a HAM n00b. I can found no documentation on this unit anywhere. Does anyone know where to start? I joined a local HAM club but have not had any time to go and pick brains. I am afraid to really even plug it in until I know what I am doing and have a call sign and everything so the FCC does't kick in my door. I did plug it in for a minute and there were no lights or anything so I not even sure it works. Anyone have any links or ideas? Thanks, Steve - Original Message - From: Don Fanning To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, September 11, 2005 1:37 PM Subject: RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM Time and time again, emergency action drills take place in cities to target where their weaknesses are in "crisis" handling. Usually they involve planes crashing or explosions (mock of course). Obviously they were never prepared for this sort of disaster in their recovery plan. I've participated in a few ARES/RACES drills and have to say that much could be done to improve upon the "HAM" infrastructure. Most of the time, communications is coordinated through 1 repeater system. When this repeater goes down, of course people would switch comms to another but in a case like this, where all the repeater systems go down except for maybe one, there needs to be a better plan. In Amateur Satellite Service, these orbiting "Repeaters" employ a system called RUDAK where a chunk of spectrum is repeated. Obviously this isn't feasible in terrestrial repeaters but they dohave the ability to turn off radios and switch bands at will depending on operating conditions. With software controlled radio and Asterisk, the repeater system could be made to be more resilient to disaster by linking to other repeater systems via radio where it could connect outward. If you figure the overhead of a repeater's transmitter and receiver plus the controller, replaceing the controller with an asterisk based unit (integration) would make more sense as it would give the repeater system much more capabilities in the same footprint and power. Additionally, these repeater systems are located on hilltops with other radio systems so they should have emergency power available (if you've ever been to a hilltop repeater site, you'll know what I mean). I think the biggest thing that hurts ham radio's ability to react to a crisis is the lack of equipment and operators. Most of the traffic we pass is "Health and Welfare" with "Logistics" being the second to it. What defeats this is that in a disaster where local/high band long haul capabilities are diminished, is simply the one repeater that is functional because everything is squeezed onto one VHF/UHF repeater. Where I could see thing being improved? Installation of 802.11b/g WLAN under Part 97. It would allow for more users into the system, there are less hardware and power components and allows the system to be dynamically configured. Asterisk could play a huge role then as it's made for IP based traffic and could re-rou
RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM
Priority traffic by ARRL standards would fall into both of these categories. What they are saying is that if someone is in a area where a ham is operating and needs to get someone hauled out via emergency services, priority traffic would take precedence over normal traffic. Not quite a Mayday situation but close to. Hams have come through for the most part but since we're way off topic, it boils down to poor planning on the emergency coordinator for a given town/county/city/state. Let's face it. When FEMA rolls in, there's no question about their communications. If they can run it through commercial terrestrial providers, fine. Otherwise, they have satellites phones that take less than a few minutes to set up (if that). Sure it's expensive to joe smith. But we're talking about the government here where justification always outweighs cost. That being said. Asterisk has tremendous value to the HAM community. People have always been happy to get a phone call from a serviceman at sea (using MARS) or using autopatches to order pizza's. I don't think that part is argued. The question is how it could be helpful? Asterisk Conferences - Add the ability for people who are HAMS to log into a protected chat room and communicate to both equipped and non equipped hams (using cell phones). Emergency services could teleconference a Public Radio Service repeater and monitor the conference to coordinate responses with lower overhead (again using COTS equipment). Asterisk Autopatching - This would allow people to setup Health and Welfare phone booths for people to call their loves ones and coordinate their return to a normal life. One feature that I see really lacking in Asterisk however is the ability to outdial from a teleconference to three-way them into a conference as well as moderator functions. Of course these features are in Alliance teleconferences but would be nice to add in as well. Cepstral Integration - Imagine if your car was stolen and it was equipped with APRS. You could write a script that would read lon/lat, do the map lookup and feed back location information every 10 seconds to assist in recovery. All it would take is 3-waying into the asterisk, logging in and having * read back the information to emergency response. The applications are endless with a system like this. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey Sent: Sunday, September 11, 2005 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM I think the biggest thing that hurts ham radio's ability to react to a crisis is the lack of equipment and operators. Most of the traffic we pass is Health and Welfare with Logistics being the second to it. You might be interested to take a listen to the latest ARRL News - they give a count of Priority traffic messages passed for Katrina... http://www.arrl.org/arrlletter/audio/ The site is ARRL and it's their ARRL Letter feed to be presented on repeaters. The ARES response to Katrina articles have the info I'm referring to. Sorry for the OT addition to the thread but I find it worth mentioning. Also, for my two cents I'll toss in that the first thing I thought of when someone mentioned using Asterisk with Ham was to get a Laptop with a WiFi connection, Asterisk and a radio interface on scene to provide comm links. 73 de NY5I Hatton Humphrey ___ --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
RE: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
Call ServiceMaster :) Depends on how much charge was left in the circuit as to what will happened. If it was saltwater, probably not. Freshwater, there might be a chance that after it dries completely that it will come back online. Won't know until you can test it. Glad you and your family is safe. I have a friend who's husband is MIA still in Gulfport. Quite a time there. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Saturday, September 03, 2005 10:05 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee Hi All, My family and I are doing well. Thank you all for your prayers. We are using this as an opportunity to rebuild. I didn't think I really needed to but God knows best and we will obey. My family and I will temporarily be in Lafayette, Louisiana for a while but will probably relocate to Houston, TX in the future. We already have my Daughter registered in school here. Lafayette is my old stomping ground so I'm already at home. My Wife is having a time with directions though. She went half way to Lake Charles (wrong direction) yesterday when she was coming back home from shopping. My house, office, lab and 2 vehicles back in Chalmette, LA, St Bernard Parish are swimming with the fishes, snakes and alligators along with all my computers and Asterisk application development. 100% loss, but hey, we have our health. I have both homeowners and flood insurance so I should recoup most of my losses, it will take a while to get back on track. Insurance adjusters will not be able to enter the Parish till the water is out which could take several weeks if not a few months. I was planning on speaking at this years Astricon conference in Anaheim, CA on Embedded Asterisk Systems but have to resend the invitation at this time. As you can imagine, I have other priorities. I will miss this opportunity to collaborate and share my work with this community. My FTP server is 8 feet under Lake Ponchatrain at this time and foreseeable future. My Internet provider is not online anyway but I am committed and will get my work on-line as soon as possible. I will keep up with Asterisk development as I can and will jump back into the community when available to contribute with focus and vigor. I have bought and collected equipment since being in Telecommunications, VoIP and Internet Technologies for 15 years that are irreplaceable but I will re-build my VoIP laboratory bigger and better than ever. If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. God Bless. JR Richardson ___ --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
RE: Re: [Asterisk-Users] VoipBuster with astersisk?
I ended up creating another account and it works great. I'll wait until
after beta and have them fix the first account.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 31, 2005 5:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re: [Asterisk-Users] VoipBuster with astersisk?
Thanks,
I'll try it. From what I read on the Internet, people start to have
problems when they pput money on their account.
They say it works ok when account is empty, but when 1euro is deposited,
client still works, but asterisk does not. Did you have any problems?
Rudolf
Mat Stace, Colewood [EMAIL PROTECTED] wrote:
I'm running voipbuster via IAX, though you'll have to change the
dialstring, as I only use it for UK landline numbers :)
In my iax.conf
[voipbuster]
type=peer
host= 213.61.187.150
secret=YOURPASSWORD
notransfer=yes
context=default
In My extensions.conf:
exten = _770[12].,1,SetCallerID(CID Name CIDNUMBER) exten =
_770[12].,2,Dial,IAX2/[EMAIL PROTECTED]/0044${EXTEN:3}
I don't actually know if the first line works (never actually tested it
that far :-| ) and you'll probably want the 2nd line to be something
like this if you want to use it for all calls worldwide
exten = _9.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:1}
This should give you the 9 for a line mdoe of operation, and require
you to dial full international numbers.
Cheers
Mat
(standard disclaimer - while the above works for me, it's for a
particular purpose. YMMV, don't sue me if it breaks, etc etc etc) ;-D
[EMAIL PROTECTED] wrote:
Hi, all
Here is a something I found on the web:
http://www.voipbuster.com
And it works OK too. Now, I want to use it via asterisk, so I ccan use
my normal phones
instead of PC application.
Did anyone try to connect astersisk and VoipBuster?
Thanks,
Rudolf
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[Asterisk-Users] Asterisk 1.2.0 fails to hang up using SIP
Greetings all, I just installed the beta now my SIP phone doesn't correctly hang up and clear trunks once the call is answered. First one is fine because it didn't answer on the far end. The second one stayed connected. -- Executing SetCallerID(SIP/100-5ab0, 516301) in new stack -- Executing Dial(SIP/100-5ab0, IAX2/[EMAIL PROTECTED]/0044289xxx) in new stack -- Called [EMAIL PROTECTED]/0044289xxx -- Call accepted by 213.61.187.147 (format ulaw) -- Format for call is ulaw -- IAX2/voipbuster2-8 is making progress passing it to SIP/100-5ab0 -- Hungup 'IAX2/voipbuster2-8' == Spawn extension (internalselections, 01144289xxx, 2) exited non-zero on 'SIP/100-5ab0' -- Executing SetCallerID(SIP/100-4740, 516301) in new stack -- Executing Dial(SIP/100-4740, IAX2/[EMAIL PROTECTED]/0044289xxx) in new stack -- Called [EMAIL PROTECTED]/0044289xxx -- Call accepted by 213.61.187.147 (format ulaw) -- Format for call is ulaw -- IAX2/voipbuster2-1 is making progress passing it to SIP/100-4740 -- IAX2/voipbuster2-1 answered SIP/100-4740 *CLI stop now Beginning asterisk shutdown -- Hungup 'IAX2/voipbuster2-1' == Spawn extension (internalselections, 01144289xxx, 2) exited non-zero on 'SIP/100-4740' Executing last minute cleanups == Destroying musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Asterisk cleanly ending (0). subspace:/etc/asterisk# ___ --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] Meetme using ztdummy on Linux 2.6 sounds scratchy
I'm currently working out the config bugs on my * box and I'm noticing that the meetme is very scratchy. As in not usable scratchy tho I can hear the audio it sounds like when you talk through a fan. Anyone have any ideas? Linux 2.6 with RTC installed. Using stable release and SIP devices. -Don ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Voipbuster blocking Asterisk/IAX connections?
VoipBuster is a service from Finarea SA Po Box 5648 Lugano 6901 CH But you are correct. The servers are supposedly housed in germany. Even accounting is the same as I couldn't get a voipcheap and a voipbuster account with the same username. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Friday, August 19, 2005 2:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? They're using the same hosted servers with different billin schemes. When I last looked there was a huge difference in ping times and voipbuster when I tested it was very much up and down in responsiveness. I thought they were in Germany (or at least Europe)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
They're using the same hosted servers with different billin schemes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Burton Sent: Tuesday, August 16, 2005 11:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? On 17 Aug 2005, at 02:26, Don Fanning wrote: I've surmized that it's Voipbuster having issues. Paid up another euro on the second account and it works fine. When their support gets better, I'll have them work on the other account. I've had similar flakyness with Voipbuster. Sometimes the call goes through a dream, next time I either get no authority found or invalid extension/context. For me it's 50/50 This seems odd.. I put it down to their free service ... [Though, what s worse, If Voipbuster fails, then voipjet fails too, in the same way, and that I REALLY dont understand! But I haven't got on that case to Voipjet yet - so i dont know what the problem is...] Cheers Mark. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Fanning Sent: Tuesday, August 16, 2005 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? I added in a second account that does not have the 1 Euro deposit and it goes through. What would make things so different? (this time the number is to the NIST Atomic Clock) --- *CLI iax2 debug IAX2 Debugging Enabled -- Executing SetCallerID(SIP/100-d2c1, jfalcon) in new stack -- Executing Dial(SIP/100-d2c1, IAX2/[EMAIL PROTECTED]/0013034997111) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 00010 DCall: 0 [213.61.187.146:4569] VERSION : 2 CALLED NUMBER : 0013034997111 CALLING NAME: jfalcon LANGUAGE: en USERNAME: jfalcon FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185631973 -- Called [EMAIL PROTECTED]/0013034997111 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00013ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 4ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] AUTHMETHODS : 3 CHALLENGE : 188826810 USERNAME: jfalcon Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00186ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] MD5 RESULT : 95fd16ba91a429b62028fc1ec6aa9cb5 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00186ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00188ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] FORMAT : 2 -- Call accepted by 213.61.187.146 (format gsm) -- Format for call is gsm Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00188ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10014ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10002ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 10729ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10729ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] -- Hungup 'IAX2/voipbuster/10' == No one is available to answer at this time -- Executing NoOp(SIP/100-d2c1, DIALSTATUS=NOANSWER) in new stack -- Executing NoOp(SIP/100-d2c1, HANGUPCAUSE=0) in new stack -- Executing Dial(SIP/100-d2c1, IAX2/[EMAIL PROTECTED]/0013034997111) in new stack Tx-Frame Retry[000
RE: [Asterisk-Users] 1-800 number
How about a sex line? :) They never pick up on those. Like 1-800-554-0069 800 numbers still charge the customer but in this case the customer is the one terminating the 800 service. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Eicke Sent: Wednesday, August 17, 2005 1:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 1-800 number Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Debug below [voipbuster] type=peer host=iax.voipbuster.com ;host=213.61.187.150 secret=x notransfer=yes context=default qualify=yes disallow=all allow=ulaw allow=alaw --- subspace*CLI iax2 debug IAX2 Debugging Enabled -- Executing SetCallerID(SIP/100-b0b3, xx) in new stack -- Executing Dial(SIP/100-b0b3, IAX2/[EMAIL PROTECTED]/1516308) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 9ms SCall: 3 DCall: 0 [213.61.187.150:4569] VERSION : 2 CALLED NUMBER : 1516308 CALLING NAME: xxx LANGUAGE: en USERNAME: xxx FORMAT : 4 CAPABILITY : 63500 ADSICPE : 2 DATE TIME : 185610345 -- Called [EMAIL PROTECTED]/1516308 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 9ms SCall: 00024 DCall: 3 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 8ms SCall: 00024 DCall: 3 [213.61.187.150:4569] AUTHMETHODS : 3 CHALLENGE : 293385486 USERNAME: xxx Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00182ms SCall: 3 DCall: 00024 [213.61.187.150:4569] MD5 RESULT : f5152720f09e919d86eeca6bb8aef5c8 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00182ms SCall: 00024 DCall: 3 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00178ms SCall: 00024 DCall: 3 [213.61.187.150:4569] FORMAT : 4 -- Call accepted by 213.61.187.150 (format ulaw) -- Format for call is ulaw Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00178ms SCall: 3 DCall: 00024 [213.61.187.150:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10010ms SCall: 3 DCall: 00024 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10010ms SCall: 00024 DCall: 3 [213.61.187.150:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10010ms SCall: 3 DCall: 00024 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10006ms SCall: 00024 DCall: 3 [213.61.187.150:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10006ms SCall: 3 DCall: 00024 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10006ms SCall: 00024 DCall: 3 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 10810ms SCall: 00024 DCall: 3 [213.61.187.150:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10810ms SCall: 3 DCall: 00024 [213.61.187.150:4569] -- Hungup 'IAX2/voipbuster/3' == No one is available to answer at this time -- Executing Congestion(SIP/100-b0b3, ) in new stack == Spawn extension (internalselections, 91516308, 3) exited non-zero on 'SIP/100-b0b3' subspace*CLI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, August 15, 2005 9:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? Don Fanning wrote: What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Um, IAX isn't encrypted, just a binary protocol. Have a look at ethereal. Also, post us your results of IAX2 Debug -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Taking in everyone's suggestions (added a username line also) here is what I got. Still no joy --- *CLI *CLI *CLI -- Executing SetCallerID(SIP/100-b225, ) in new stack -- Executing Dial(SIP/100-b225, IAX2/[EMAIL PROTECTED]/001516308) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 6 DCall: 0 [213.61.187.157:4569] VERSION : 2 CALLED NUMBER : 001516308 CALLING NAME: x LANGUAGE: en USERNAME: x FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185630028 -- Called [EMAIL PROTECTED]/001516308 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00015ms SCall: 00325 DCall: 6 [213.61.187.157:4569] AUTHMETHODS : 3 CHALLENGE : 203796716 USERNAME: xx Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00191ms SCall: 6 DCall: 00325 [213.61.187.157:4569] MD5 RESULT : e682d22660c7a0d278bef6025bcc7dc0 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00191ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00183ms SCall: 00325 DCall: 6 [213.61.187.157:4569] FORMAT : 2 -- Call accepted by 213.61.187.157 (format gsm) -- Format for call is gsm Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00183ms SCall: 6 DCall: 00325 [213.61.187.157:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00010ms SCall: 7 DCall: 0 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00010ms SCall: 00088 DCall: 7 [213.61.187.157:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00010ms SCall: 7 DCall: 00088 [213.61.187.157:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10017ms SCall: 6 DCall: 00325 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10017ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10017ms SCall: 6 DCall: 00325 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10009ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10009ms SCall: 6 DCall: 00325 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10009ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 10948ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10948ms SCall: 6 DCall: 00325 [213.61.187.157:4569] -- Hungup 'IAX2/voipbuster/6' == No one is available to answer at this time -- Executing Congestion(SIP/100-b225, ) in new stack == Spawn extension (internalselections, 9001516308, 3) exited non-zero on 'SIP/100-b225' *CLI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Tuesday, August 16, 2005 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? On 8/16/05, Tony Hoyle [EMAIL PROTECTED] wrote: Don Fanning wrote: CALLED NUMBER : 1516308 Is that a valid number? AFAIK all voipbuster numbers have to start with 0 as there's no local dialing. Assuming that number is a US number, area code 516, it should be dialed as 001516308. Number format at Voipbuster is 00 country area number -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be lamitated! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Btw: The number is to my stanaphone DID (so it doesn't bug anyone) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Fanning Sent: Tuesday, August 16, 2005 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? Taking in everyone's suggestions (added a username line also) here is what I got. Still no joy --- *CLI *CLI *CLI -- Executing SetCallerID(SIP/100-b225, ) in new stack -- Executing Dial(SIP/100-b225, IAX2/[EMAIL PROTECTED]/001516308) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 6 DCall: 0 [213.61.187.157:4569] VERSION : 2 CALLED NUMBER : 001516308 CALLING NAME: x LANGUAGE: en USERNAME: x FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185630028 -- Called [EMAIL PROTECTED]/001516308 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00015ms SCall: 00325 DCall: 6 [213.61.187.157:4569] AUTHMETHODS : 3 CHALLENGE : 203796716 USERNAME: xx Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00191ms SCall: 6 DCall: 00325 [213.61.187.157:4569] MD5 RESULT : e682d22660c7a0d278bef6025bcc7dc0 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00191ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00183ms SCall: 00325 DCall: 6 [213.61.187.157:4569] FORMAT : 2 -- Call accepted by 213.61.187.157 (format gsm) -- Format for call is gsm Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00183ms SCall: 6 DCall: 00325 [213.61.187.157:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00010ms SCall: 7 DCall: 0 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00010ms SCall: 00088 DCall: 7 [213.61.187.157:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00010ms SCall: 7 DCall: 00088 [213.61.187.157:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10017ms SCall: 6 DCall: 00325 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10017ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10017ms SCall: 6 DCall: 00325 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10009ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10009ms SCall: 6 DCall: 00325 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10009ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 10948ms SCall: 00325 DCall: 6 [213.61.187.157:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10948ms SCall: 6 DCall: 00325 [213.61.187.157:4569] -- Hungup 'IAX2/voipbuster/6' == No one is available to answer at this time -- Executing Congestion(SIP/100-b225, ) in new stack == Spawn extension (internalselections, 9001516308, 3) exited non-zero on 'SIP/100-b225' *CLI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Tuesday, August 16, 2005 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? On 8/16/05, Tony Hoyle [EMAIL PROTECTED] wrote: Don Fanning wrote: CALLED NUMBER : 1516308 Is that a valid number? AFAIK all voipbuster numbers have to start with 0 as there's no local dialing. Assuming that number is a US number, area code 516, it should be dialed as 001516308. Number format at Voipbuster is 00 country area number -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be lamitated! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Done
---
*CLI
*CLI
*CLI
-- Executing SetCallerID(SIP/100-1ba9, x) in new stack
-- Executing Dial(SIP/100-1ba9,
IAX2/[EMAIL PROTECTED]/001516308) in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 8ms SCall: 00010 DCall: 0 [213.61.187.157:4569]
VERSION : 2
CALLED NUMBER : 001516308
CALLING NAME: x
LANGUAGE: en
USERNAME: x
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185630951
-- Called [EMAIL PROTECTED]/001516308
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 8ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 1ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
AUTHMETHODS : 3
CHALLENGE : 229696652
USERNAME: x
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00180ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
MD5 RESULT : 8b729ab88c50ba655fef99ef151ad228
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00180ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00171ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
FORMAT : 2
-- Call accepted by 213.61.187.157 (format gsm)
-- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00171ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10009ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10051ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10051ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10009ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10009ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10051ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
HANGUP
Timestamp: 10934ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Unknown IE 042 : Present
Ignoring unknown information element 'Unknown IE' (42) of length 1
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass:
ACK
Timestamp: 10934ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
-- Hungup 'IAX2/voipbuster/10'
== No one is available to answer at this time
-- Executing NoOp(SIP/100-1ba9, DIALSTATUS=NOANSWER) in new
stack
-- Executing NoOp(SIP/100-1ba9, HANGUPCAUSE=0) in new stack
-- Executing Congestion(SIP/100-1ba9, ) in new stack
== Spawn extension (internalselections, 9001516308, 6) exited
non-zero on 'SIP/100-1ba9'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling aka ManxPower
Sent: Tuesday, August 16, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX
connections?
Don Fanning wrote:
Taking in everyone's suggestions (added a username line also) here is
what I got.
Still no joy
---
*CLI
*CLI
*CLI
-- Executing SetCallerID(SIP/100-b225, ) in new stack
-- Executing Dial(SIP/100-b225,
IAX2/[EMAIL PROTECTED]/001516308) in new stack
-- Called [EMAIL PROTECTED]/001516308
-- Hungup 'IAX2/voipbuster/6'
== No one is available to answer at this time
-- Executing Congestion(SIP/100-b225, ) in new stack
== Spawn extension (internalselections, 9001516308, 3) exited
non-zero on 'SIP/100-b225'
Put a Noop(HANGUPCAUSE=${HANGUPCAUSE}) and a
Noop(DIALSTATUS=${DIALSTATUS}) as the two priorities after your Dial to
see WHY the call was hungup.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. r makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00222ms SCall: 5 DCall: 00148 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: (14?) Timestamp: 02557ms SCall: 00148 DCall: 5 [213.61.187.147:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 02557ms SCall: 5 DCall: 00148 [213.61.187.147:4569] -- IAX2/voipbuster2/5 is making progress passing it to SIP/100-d2c1 Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 003 Type: VOICE Subclass: 2 Timestamp: 02800ms SCall: 5 DCall: 00148 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: VOICE Subclass: 2 Timestamp: 02688ms SCall: 00148 DCall: 5 [213.61.187.147:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02688ms SCall: 5 DCall: 00148 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 02800ms SCall: 00148 DCall: 5 [213.61.187.147:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 8ms SCall: 1 DCall: 0 [213.61.187.147:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00011ms SCall: 6 DCall: 0 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 8ms SCall: 00286 DCall: 1 [213.61.187.147:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 8ms SCall: 1 DCall: 00286 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00011ms SCall: 00342 DCall: 6 [213.61.187.146:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00011ms SCall: 6 DCall: 00342 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRQ Timestamp: 10018ms SCall: 5 DCall: 00148 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10018ms SCall: 00148 DCall: 5 [213.61.187.147:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10018ms SCall: 5 DCall: 00148 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: LAGRQ Timestamp: 10017ms SCall: 00148 DCall: 5 [213.61.187.147:4569] Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 006 Type: IAX Subclass: LAGRP Timestamp: 10017ms SCall: 5 DCall: 00148 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10017ms SCall: 00148 DCall: 5 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 005 Type: CONTROL Subclass: ANSWER Timestamp: 14131ms SCall: 00148 DCall: 5 [213.61.187.147:4569] Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 007 Type: IAX Subclass: ACK Timestamp: 14131ms SCall: 5 DCall: 00148 [213.61.187.147:4569] -- IAX2/voipbuster2/5 answered SIP/100-d2c1 -- Hungup 'IAX2/voipbuster2/5' == Spawn extension (internalselections, 90013034997111, 5) exited non-zero on 'SIP/100-d2c1' Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 007 Type: IAX Subclass: HANGUP Timestamp: 19399ms SCall: 5 DCall: 00148 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 19399ms SCall: 00148 DCall: 5 [213.61.187.147:4569] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Fanning Sent: Tuesday, August 16, 2005 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? Done --- *CLI *CLI *CLI -- Executing SetCallerID(SIP/100-1ba9, x) in new stack -- Executing Dial(SIP/100-1ba9, IAX2/[EMAIL PROTECTED]/001516308) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 8ms SCall: 00010 DCall: 0 [213.61.187.157:4569] VERSION : 2 CALLED NUMBER : 001516308 CALLING NAME: x LANGUAGE: en USERNAME: x FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185630951 -- Called [EMAIL PROTECTED]/001516308 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 8ms SCall: 00306 DCall: 00010 [213.61.187.157:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 1ms SCall: 00306 DCall: 00010 [213.61.187.157:4569] AUTHMETHODS : 3 CHALLENGE : 229696652 USERNAME: x Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00180ms SCall
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
I've surmized that it's Voipbuster having issues. Paid up another euro on the second account and it works fine. When their support gets better, I'll have them work on the other account. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Fanning Sent: Tuesday, August 16, 2005 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? I added in a second account that does not have the 1 Euro deposit and it goes through. What would make things so different? (this time the number is to the NIST Atomic Clock) --- *CLI iax2 debug IAX2 Debugging Enabled -- Executing SetCallerID(SIP/100-d2c1, jfalcon) in new stack -- Executing Dial(SIP/100-d2c1, IAX2/[EMAIL PROTECTED]/0013034997111) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 00010 DCall: 0 [213.61.187.146:4569] VERSION : 2 CALLED NUMBER : 0013034997111 CALLING NAME: jfalcon LANGUAGE: en USERNAME: jfalcon FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185631973 -- Called [EMAIL PROTECTED]/0013034997111 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00013ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 4ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] AUTHMETHODS : 3 CHALLENGE : 188826810 USERNAME: jfalcon Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00186ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] MD5 RESULT : 95fd16ba91a429b62028fc1ec6aa9cb5 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00186ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00188ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] FORMAT : 2 -- Call accepted by 213.61.187.146 (format gsm) -- Format for call is gsm Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00188ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10014ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10002ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 10729ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10729ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] -- Hungup 'IAX2/voipbuster/10' == No one is available to answer at this time -- Executing NoOp(SIP/100-d2c1, DIALSTATUS=NOANSWER) in new stack -- Executing NoOp(SIP/100-d2c1, HANGUPCAUSE=0) in new stack -- Executing Dial(SIP/100-d2c1, IAX2/[EMAIL PROTECTED]/0013034997111) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00018ms SCall: 5 DCall: 0 [213.61.187.147:4569] VERSION : 2 CALLED NUMBER : 0013034997111 CALLING NAME: jfalcon LANGUAGE: en USERNAME: jfalcontwo FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185631979 -- Called [EMAIL PROTECTED]/0013034997111 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00018ms SCall: 00148 DCall: 5 [213.61.187.147:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00052ms SCall: 00148 DCall: 5 [213.61.187.147:4569] AUTHMETHODS : 3 CHALLENGE : 529526436 USERNAME: jfalcontwo Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00227ms SCall: 5 DCall: 00148 [213.61.187.147:4569] MD5
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Any assistance? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Hoyle Sent: Monday, August 15, 2005 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? Erick Weber V. wrote: For me to Works for me... Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
That's what I have as well... What codec are you running with connections to it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Hoyle Sent: Monday, August 15, 2005 8:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? Don Fanning wrote: What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. [voipbuster] host=iax.voipbuster.com type=peer username=xx secret=x qualify=yes context=inbound Nothing special... And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. iax isn't encrypted Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Will do as soon as I bring it back online (took it off yesterday to do some housecleaning on the server). Maybe that's what I'm mistaking it for. I didn't see a clear channeled header for communicating. The typical error I would get is that nobody available/no answer after it connects the IAX circuit. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, August 15, 2005 9:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? Don Fanning wrote: What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Um, IAX isn't encrypted, just a binary protocol. Have a look at ethereal. Also, post us your results of IAX2 Debug -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using prepaid calling cards to dial out with Asterisk - extensions.conf
Title: Using prepaid calling cards to dial out with Asterisk - extensions.conf How would one setup a extensions.conf (or other file) to use calling cards to dial out with? Thanks, Don Fanning Freelance Hacker - Producer of the 3 M's (Music, Movies and Microcode) Wherever you go, There you are. - Buckaroo Banzai ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX2 and FWD - Wrong context?
Title: IAX2 and FWD - Wrong context? Greets all, I'm setting up asterisk and trying to get IAX2 running for FWD. I followed the FWD IAX2 page verbatim but I get the following error May 14 08:09:31 WARNING[7569]: chan_iax2.c:5569 socket_read: Call rejected by 65.39.205.121: No such context/extension The rsa key is in, that's the only error I'm seeing when I try calling out with this. I haven't tried inbound yet. Any help would be appreciated. Thanks, Don Fanning Freelance Hacker - Producer of the 3 M's (Music, Movies and Microcode) Wherever you go, There you are. - Buckaroo Banzai ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX2 and FWD - Wrong context?
to the demo site
exten = 500,4,Goto(s,6); Return to the start over message.
;
; Create an extension, 600, for evaulating echo latency.
;
exten = 600,1,Playback(demo-echotest) ; Let them know what's going on
exten = 600,2,Echo ; Do the echo test
exten = 600,3,Playback(demo-echodone) ; Let them know it's over
exten = 600,4,Goto(s,6); Start over
;
; Give voicemail at extension 8500
;
exten = 0,1,VoicemailMain
exten = 0,2,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten = 1265,1,Dial(Phone/phone0,15)
;exten = 1265,2,Goto(s,5)
;123 plays the time
exten = 123,1,Answer
exten = 123,2,SayUnixTime()
exten = 123,3,Hangup
;321 plays a stream
exten = 321,1,Answer
exten = 321,2,MP3Player,http://64.236.34.97:80/stream/1003
;890 plays local music file
exten = 890,1,Answer()
exten = 890,2,Wait,1
exten = 890,3,mp3player(/var/lib/asterisk/music/890_hold.mp3)
exten = 890,4,Goto(890,1)
;503 is a key test
exten = 503,1,Playback(digits/h-1)
exten = 503,2,Wait(1)
exten = 503,3,Playback(digits/mon-1)
exten = 503,4,Wait(1)
exten = 503,5,Playback(digits/oh)
exten = 503,6,Hangup
[macro-stdexten];
exten = s,1,GotoIf($[${ARG1} = 10003]?30:2) ; is it a call for don? if yes
continue if no goto 30
;save all the caller id info (for 1471) in CALLOG/EXTEN
exten = s,2,DBput(CALLLOG/${ARG1}=${CALLERIDNUM})
exten = s,3,DBput(CALLLOG/${ARG1}-month=${TIMESTAMP:4:2})
exten = s,4,DBput(CALLLOG/${ARG1}-day=${TIMESTAMP:6:2})
exten = s,5,DBput(CALLLOG/${ARG1}-hour=${TIMESTAMP:9:2})
exten = s,6,DBput(CALLLOG/${ARG1}-minute=${TIMESTAMP:-4:2})
exten = s,7,DBput(CALLLOG/${ARG1}-second=${TIMESTAMP:-2:2})
;play transfer (please hold while I connect you)
exten = s,8,Playback(transfer)
;Ring ring, call arg2 (but put caller on a Music on hold process)
exten = s,9,Dial(${ARG2},25,m)
;If unavailable, send to voicemail w/ unavail announce
exten = s,10,Voicemail(u${ARG1})
exten = s,11,Hangup()
exten = s,102,Voicemail(b${ARG1})
exten = s,103,Hangup()
;adds callid stuff to everyone's extension
exten = s,30,DBput(CALLLOG/10001=${CALLERIDNUM})
exten = s,31,DBput(CALLLOG/10001-month=${TIMESTAMP:4:2})
exten = s,32,DBput(CALLLOG/10001-day=${TIMESTAMP:6:2})
exten = s,33,DBput(CALLLOG/10001-hour=${TIMESTAMP:9:2})
exten = s,34,DBput(CALLLOG/10001-minute=${TIMESTAMP:-4:2})
exten = s,35,DBput(CALLLOG/10001-second=${TIMESTAMP:-2:2})
exten = s,36,DBput(CALLLOG/10002=${CALLERIDNUM})
exten = s,37,DBput(CALLLOG/10002-month=${TIMESTAMP:4:2})
exten = s,38,DBput(CALLLOG/10002-day=${TIMESTAMP:6:2})
exten = s,39,DBput(CALLLOG/10002-hour=${TIMESTAMP:9:2})
exten = s,40,DBput(CALLLOG/10002-minute=${TIMESTAMP:-4:2})
exten = s,41,DBput(CALLLOG/10002-second=${TIMESTAMP:-2:2})
;goto 8
exten = s,42,Goto(8)
[mainmenu]
;
;the main menu
;
;wait two seconds (to let the 2nd ring come through (for bellcore cid))
exten = s,1,Wait,1
exten = s,2,Wait,2
;answer it
exten = s,3,Answer
;play welcome message
exten = s,4,Background(hello)
;play end message
exten = s,5,Background(end)
;play the short 890 (20 sec)
exten = s,6,Background(890)
;play hello again
exten = s,7,Background(hello)
;play end again
exten = s,8,Background(end)
;play 5 minute 890
exten = s,9,Background(890long)
;wtf are they doing sitting there for 5 minutes
exten = s,10,Background(realend)
;hangup
exten = s,11,Hangup
;press 1 for Don
exten = 1,1,Macro(stdexten,10001,${DON})
exten = don,1,Goto(10001|1)
;press 2 for Alan
exten = 2,1,Macro(stdexten,10002,${ALAN})
exten = alan,1,Goto(10002|1)
Thanks,
Don Fanning
Freelance Hacker - Producer of the 3 M's (Music, Movies and Microcode)
Wherever you go, There you are. - Buckaroo Banzai
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wilson Pickett
Sent: Saturday, May 14, 2005 1:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX2 and FWD - Wrong context?
I'm setting up asterisk and trying to get IAX2 running for FWD. I
followed the FWD IAX2 page verbatim but I get the following error.
May 14 08:09:31 WARNING[7569]: chan_iax2.c:5569 socket_read: Call
rejected by 65.39.205.121: No such context/extension
How about giving us a look at your dial exten ?
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