Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling
On Feb 23, 2008, at 06:52 , Yehavi Bourvine +972-8-9489444 wrote: Hello, I've one nokia E65 that works very well with my asterisk box. The people here don't let me even try it as they are afraid it will consume the battery more than when it is used the usual way. Is this true? Yes, this is very true. Keeping WLAN active to stay connected to the SIP server means atrocious battery life. At least on my E60. At this point I get maybe 30 hours out of a charge when I use 30-60 minutes speaking time on the phone. jens ___ -- 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] Info: Nokia E65 working with Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 8 Mar 2007, at 13:34, Olivier wrote: I have left the default for outgoing calls to be the mobile network. To make a call via the Asterisk PBX, you need to enter the number then press the 'options' key, select 'Call' go to 'Internet Call'. Is this 'Call' go to 'Internet Call' usable when you select a callee using the phone's directory ? Yes it is. However, this also depends on how you set up your dial plan and how you store phone numbers in your directory. I have set up my Asterisk dial plan to understand and work with the universal phone number notation of +country codearea codenumber, which is understood by the mobile network as well. I store all my phone numbers that way, be they local, long distance or international long distance from where I am. This means I can select any phone number from my phone book and dial out via the mobile network or my Asterisk server, it just works. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF8AnFRAx5nvEhZLIRAqPbAKCH2IxZAvTTtt4D8WjbzU5WVz6FGACfTVD6 bAaLd67dNaiatajZ3nSdP4A= =V36x -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Sending SMS
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 3 Mar 2007, at 15:02, Steve Totaro wrote: Text messaging is not that big in the US for some reason. Well anyways, on my T-Mobile phone, I have an unlimited text message package that cost $15/mo. I am not sure how many constitutes unlimited though, I have not read the small print. If texting were as popular in the US they would not have unlimited tariffs, they're still trying to get you hooked so you can pay through the nose later on ;) jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF6a3fRAx5nvEhZLIRAkHQAJ0VyGNbKVQnVBH2yZnr5sN/4GxyUACfdaGb fz7ZvWg0RKgqOOeHYozeepY= =qaGg -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 22 Feb 2007, at 07:25, Tzafrir Cohen wrote: I tried to use the 1.2.x RPMs and they would not work for me attempting to use them with an Eicon Diva Server card and Melware's chan_capi. Only by looking at the SRPM did I notice that they are patched with BRIStuff patches, which I have assume causes incompatibilities. Why is the a problem? The bristuff zaptel patch is a really small and non-intrussive one. The bristuff Asterisk patch, though, includes a complete reimplementation of chan_capi (the Junghanns' original chan_capi), which I heard noone really uses. Specifically, some simple AGI script I run to send and receive faxes with chan_capi did not work anymore. Both you and Axel are right about rebuilding the RPM of course. Matter of fact I always strongly prefer packages that come from (trusted) yum repositories. However, in this special case if I have to rebuild the package every time I don't see much advantage over a standard source install, which is very quick and simple. I just don't want to spend time analyzing a spec file to see which patches are applied and, if needed, back them out and build again and see if my stuff works again. It would be worth it if I had more than a single server running Asterisk. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF3VEDRAx5nvEhZLIRAgcTAJ0YXMtebIMdeuGPJ4rr3yilbEYDrgCgowKp 6Or+CuV7NIxLIGVp/ApIwHs= =J7Fe -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 21 Feb 2007, at 23:06, Axel Thimm wrote: On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote: I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64 very good, but since FC keeps updating, I tried to follow newer kernel versions. If you want to save these hassles, why not use the packages bits that are available for FC5/FC6/RHEL4/RHEL3 i386/x86_64/ppc? There are even packages for the upcoming F7 and RHEL5 available: Hi Axel, I tried to use the 1.2.x RPMs and they would not work for me attempting to use them with an Eicon Diva Server card and Melware's chan_capi. Only by looking at the SRPM did I notice that they are patched with BRIStuff patches, which I have assume causes incompatibilities. Compiling Asterisk and Zaptel from sources again solved all my problems. It may be helpful to spell out more clearly how severaly patched the Asterisk in those RPMs is. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF3Ns2RAx5nvEhZLIRAtmSAJ4/ANMLSgUITOSaITMlxHhxJO1s7ACgjic6 zvPjhF6GkAvTW83JqJOtht0= =s08b -END PGP SIGNATURE- ___ --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] moving WiFi phone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 19 Feb 2007, at 15:01, Noah Miller wrote: The WAP54's have a 'repeater' mode which I've used on occasion. Which is all well and good, but they use WDS which doesn't work with WPA. Not on the WAP54's anyway (I learned the hard way on that one). Some vendors have working solutions: Apple Airports do WDS and WPA/WPA2 just fine. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF2dEBRAx5nvEhZLIRAnYYAJ0W9Bc+yredI/++EQgUPwvSDBLtXACgl5rp f/tzrwHxlf6Me8MVx1H7l4k= =Wz5J -END PGP SIGNATURE- ___ --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] moving WiFi phone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 15 Feb 2007, at 01:39, Leo Ann Boon wrote: Bruce Reeves wrote: In my experience having ap's with the same SSID and 3 channels of separation overlapping worked if the phone could roam. Recommended is 5 channels of separation. Ronald, Just be aware that even if the phone supports AP roaming, there's no guarantee that the call will continue smoothly from AP to AP. In some cases, it might take a few seconds to handover. I have two APs (Apple AirPorts) sending on the _same_ channel. Handover works perfect with no discernible loss of connectivity or audio using a Siemens SL75. The handover cannot even be noticed. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF1CK+RAx5nvEhZLIRAgJgAJ9rXMM7xQuQNaYCdUSziFz0UVbE4ACfdSuH FeEFtrmJttLNUBdrIi8DuTU= =jTGX -END PGP SIGNATURE- ___ --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] moving WiFi phone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 15 Feb 2007, at 10:23, Pavel Jezek wrote: Jens Vagelpohl wrote: I have two APs (Apple AirPorts) sending on the _same_ channel. Handover works perfect with no discernible loss of connectivity or audio using a Siemens SL75. The handover cannot even be noticed. as I know, best practice says, that neighboring AP should use _non overlapping_ channels... :-\ works for me is all I can say. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF1C04RAx5nvEhZLIRAtuKAJ94ZKW0/WZkPnoM9hUQm+hHAJ+5cACgtir5 1fRums89u32Kleaf0fCuP+Y= =IFN9 -END PGP SIGNATURE- ___ --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] Ebay Unwired Buyer, Using Asterisk?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 19 Jan 2007, at 21:07, Steve Totaro wrote: Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). There was a link posted to an interview with Allison a few weeks back. She mentioned eBay as a customer, and how she used eBay unwired before and and listened to herself speak. It doesn't mean they use Asterisk, though. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFFsSoDRAx5nvEhZLIRAojzAJwKfbZGsuFQO45ds0+ZY0jh4wYhawCfT5q1 MT40p83x78dm0CIxQGpNh8c= =4aFS -END PGP SIGNATURE- ___ --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] About BRI / ISDN hardware. What to buy?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 18 Jan 2007, at 18:31, Patrick wrote: I think http://www.melware.de carries the Eicon Server ISDN cards which have hardware echo cancellation. They are also the author of the chan_capi driver for Asterisk. I use the Eicon Server BRI cards with Asterisk myself and they work very well. I concur, I have a Eicon DIVA single port BRI card and it works very well. Cosmin, if you want to use it for Fax traffic as well make sure you do *not* get a V-BRI card. Those will not do Fax. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFFr7NLRAx5nvEhZLIRAlzZAKCcyVqEB1PcekFmFq04gJ1IjiK36QCfZQ26 8PZj2V5wU201Eu/+U/W1ihM= =NwKd -END PGP SIGNATURE- ___ --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] About BRI / ISDN hardware. What to buy?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 18 Jan 2007, at 19:49, Cosmin Prund wrote: I finally found a price tag for the darn thing, at around 500 euros I can handle it. Qustion: Do they behave properly if I've got an other Digium TDM400 card in the system? How about installing two cards in the same server? At the moment I've only got 1 ISDN line plus a few analog lines going into the TDM but in the very near future we might want to get a second ISDN. I only run a single card in my system because I don't need more ;) jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFFr8w/RAx5nvEhZLIRAmfNAKCaPOVpBopQ6bm9ji7s3290qhuHewCfcALv gE+sPbWo2N7ElOGRWeHVBi0= =hzw0 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 30 Dec 2006, at 04:59, Vernier Umali wrote: I do not have any luck using nokia E61 (doesn't register and keeps on hanging). I would think it's the same with all wifi enabled nokias. Sweeping generalizations never work. My E60 works fantastic with my Asterisk server. The sound quality is much better even than a normal cordless analog phone connected to an iaxy on my desk. With the recommendations on this thread I ordered the Siemens cordless WIFI phone to replace that iaxy/analog phone setup since the dang thing just dies on me so often. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFFljHQRAx5nvEhZLIRAimcAJ4x7SVSJYaSQY1CLwkJAjmaiYK++wCfR96T n3hgUO+luT5FacLQ2BCA3FU= =gZcp -END PGP SIGNATURE- ___ --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] Busy signal from IAXy when not connecting to my Asterisk box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 27 Nov 2006, at 19:57, Frank Tarczynski wrote: I'm having a problem with my IAXy not always connecting to my Asterisk box. When I pick-up the phone plugged in to the IAXy I get a busy signal. I have to hang-up the phone and wait a few seconds after the orange LED goes out and then try again. Very same situation here. And when someone calls me the iaxy- connected phone does not ring, Asterisk signals congestion. Sometimes I need to power down the iaxy to make it work again. I'm also using a Nokia E60 that connects to the same Asterisk server, and it's a bit sad to say that the E60, warts and all, is a lot more reliable in terms of staying connected to Asterisk. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFFa3ZoRAx5nvEhZLIRAhidAJ9Y40wcKyAW9IgFLmKCfv2sg2bi9wCeMoPa /euF4OoA1hpwkhIiswz0Cn4= =CXRO -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending fax with chan-capi
How can I generate sff format? I found sfftobmp, not nothing the other way round. You can use ghostscript: gs -dNOPAUSE -dBATCH -sDEVICE=cfax -sOutputFile=output.sff input.ps Is there a nice way to get the sff out of an Windows application (like virtual printers for hylafax) or at least some scripts which produce the sff and the asterisk call file out of an pdf? Here's something I use (not Windoze, sorry): http://svn.dataflake.org/filedetails.php? repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts% 2Ffile2fax.pyrev=0sc=0 The script takes TIFF, PS or PDF as input, creates SFF and a call file. It is run out of cron and checks if suitable files have been dropped into a spool directory. The whole package at http://svn.dataflake.org/listing.php? repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts% 2Frev=0sc=0 contains some documentation and also a script that I use to handle incoming faxes (with capicommand receivefax). jens ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending fax with chan-capi
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Klaus, The incoming fax script will generate an email with the fax attached, and there is another script, sendfax_status.py, which is run as a DeadAGI after the outgoing fax has been sent, it retrieves status information and sends it to a (hardcoded) email address. The call file created by the outgoing script file2fax.py specifies 3 retries in case of failure. This just retries it within Asterisk, I don't know if I could have chan_capi do that. jens On 11 Oct 2006, at 09:52, Klaus Darilion wrote: Hi Jens! Thanks for the script. Do you generate and notifications (succeeded, failed) or retransmit in case of failed sending? Or does that CAPI internally? regards klaus Jens Vagelpohl wrote: How can I generate sff format? I found sfftobmp, not nothing the other way round. You can use ghostscript: gs -dNOPAUSE -dBATCH -sDEVICE=cfax -sOutputFile=output.sff input.ps Is there a nice way to get the sff out of an Windows application (like virtual printers for hylafax) or at least some scripts which produce the sff and the asterisk call file out of an pdf? Here's something I use (not Windoze, sorry): http://svn.dataflake.org/filedetails.php? repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts% 2Ffile2fax.pyrev=0sc=0 The script takes TIFF, PS or PDF as input, creates SFF and a call file. It is run out of cron and checks if suitable files have been dropped into a spool directory. The whole package at http://svn.dataflake.org/listing.php? repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts% 2Frev=0sc=0 contains some documentation and also a script that I use to handle incoming faxes (with capicommand receivefax). jens ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFFLQ6qRAx5nvEhZLIRAh3SAKCBt6XOf98C2IfoPjkIGms8AbTO3ACglmU5 iyx3xR0dijuk0VnrK3bggCg= =/XV9 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: sending fax with chan-capi
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 11 Oct 2006, at 13:19, Stefan Tichy wrote: On Wed, Oct 11, 2006 at 11:32:57AM -0400, Jens Vagelpohl wrote: The call file created by the outgoing script file2fax.py specifies 3 retries in case of failure. Fax may fail even if the phone call was successfull. This just retries it within Asterisk, I don't know if I could have chan_capi do that. chan_capi 0.7 does set some variables which can / should be used in the dialplan (FAXSTATUS, ) You're absolutely right, I just haven't had the time to make this better :) jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFFLW2ARAx5nvEhZLIRAnmsAJ9qzXq5zvlLbAJ+u4ZuJQn8n3f7xgCggszk LyV8ZngayBUea0FyF8+KDzE= =h+QA -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: SV: E61
On 27 Aug 2006, at 04:03, Dovid Bender wrote: I was not going to get it based on what people said about the E61 and the NAT issues. Is this false ? I was thinking of getting it for when I travel to Israel. There seems to be a lot of open wifi connections all over the country there. Also how is the radio for the wifi on it ? Speaking for the E60, the Wifi radio is kind of shitty. It likes to just disconnect out of the blue and will only reconnect if the phone is rebooted. When the connection is up you can get decent download speeds and SIP calls are crystal clear, though ;) I never tried to use the SIP client through NAT. My Asterisk is on the same LAN. jens ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: SV: E61
On 26 Aug 2006, at 07:57, Martin Joseph wrote: Now, the fact you can't easily get these phone in the US, that's a conspiracy ;~) ... and if you take them to the US you realize you should have gotten a quad-band phone because your E60 can't deal with the common US frequency of 850 MHz, which European tri-bands don't have. My reception is bad pretty much wherever I am, using Cingular :( jens ___ --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] Now that Nufone is dead...
On 5 Jul 2006, at 19:00, John Kington wrote: At 09:29 AM 7/5/2006 +0300, you wrote: I have tollfree numbers with Nufone working OK. But what I like most is the regular numbers with charge/month but no charge/min on incoming calls... Did your tollfree number(s) with Nufone get cut-off in April? Did you keep the same number or did you signup for another number? I requested Nufone transfer my tollfree number in May and it is still not working (code is 77-4). I am wondering if this has happened to everyone or if my number fell through the cracks. I kept my toll-free number, it just took an extra step to get it working. Apparently their old provider is screwing them and taking whatever time they want to do the porting. To speed it up there's an option called force-porting, as they explained to me. The old provider will act faster for a fee of $60. NuFone is willing to eat that cost if you put those $60 into your NuFone account and put For QUICK TFN port in the payment comments, they will initiate the force porting request but leave that amount in your account so you can use it for your normal call activity. Pretty fair I'd say. The only problem is that they might not have advertised this option to every customer stuck with that particular problem. Other than that, any new toll-free DID as well as standard DID you order works instantly. jens ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?
On 4 Jul 2006, at 09:58, Olle E Johansson wrote: I've had a lot of issues with the Nokia loosing the registration and WLAN access while I'm still in the office. Anyone that have any remedies for that? Yep, that's my main issue as well. I doubts it's a configuration issue since there isn't all that much to configure. Maybe a software upgrade on the phone will help - apparently there has been one small upgrade since the version on my phone (1.0610.02.15), although for a different issue. jens ___ --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: SV: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?
On 3 Jul 2006, at 09:30, Amund Nygaard wrote: Seems the E series is suited for voip over sip. I have testet E60 my self with asterisk, at it works well. The E60, at least for me, has a fatal flaw: After connecting to the access point just fine for a while (could be hours, or even a couple days) it just disconnects without any apparent reason, and the only way to make it reconnect is to reboot the phone. If it could stay connected it would have the potential to replace all standard telephones at my place. Amund, since you work with Nokia you might know this: Is there a website somewhere that shows what the latest Nokia software versions are? Sounds like my problem could be fixed by a software update. jens ___ --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: SV: SV: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?
On 3 Jul 2006, at 09:57, Amund Nygaard wrote: As far as i know there are only support sites for the service center. I can try and look it up. Mine has 1.0610.02.15 Mine has the same version. If you could double-check that this is the latest version that would be great! jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Don't use CDRTool From AG-projescts
On 22 Jun 2006, at 22:11, Christian Stredicke wrote: This post cannot be left without comment. People who don't know you or Adrian might get a wrong impression. Honestly, I think it can. That post tells you everything you need to know about the camplaining party ;) jens ___ --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] Small form factor system w/PCI slot
On 9 Jun 2006, at 02:04, Leo Ann Boon wrote: Jens Vagelpohl wrote: Hi everyone, I'm trying to buy a small form-factor PC system for use with Asterisk and Hylafax in conjunction with a Eicon DIVA Server single-port ISDN card (needs full-size 5V PCI 2.2 slot, but PCI-X compatible). Use is very light - at most a single call at any one time. If the Mac Mini had a PCI slot I'd try to use that one, but oh well ;) You mean PCI-E? If you really need PCI-X, then you're out of luck. PCI-X is only available on server boards. For a single port ISDN, one of those Mini-ITX boxes should work. I built something similar using a Mini-ITX (1GHz CPU) with an AVM Fritz! PCI ISDN card using chan_capi. IIRC, Xorcom has a TS-1 which is a SFF Asterisk server for $500. BTW, I don't think the Mini-ITX mobos can support PCI-E. It's a normal 5.5 V PCI slot, the card can also deal with PCI-X slots as the documentation claims. I'll take a look at Xorcom's offerings, thanks. jens ___ --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] Small form factor system w/PCI slot
Hi everyone, I'm trying to buy a small form-factor PC system for use with Asterisk and Hylafax in conjunction with a Eicon DIVA Server single-port ISDN card (needs full-size 5V PCI 2.2 slot, but PCI-X compatible). Use is very light - at most a single call at any one time. If the Mac Mini had a PCI slot I'd try to use that one, but oh well ;) Would anyone have some brand names or sites where such systems can be found? I'm not really interested in building one from scratch, I'd rather buy a complete system and just stick the ISDN card in there. Thanks :) jens ___ --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] Now that Nufone is dead...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 23 May 2006, at 15:48, Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? Nufone is not dead, works perfectly fine for me. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEcybORAx5nvEhZLIRAnAoAJwJ0Ig4EUdrfw1RhTe8ULxzzq3dQQCfesYc +U9WV0uDc/qD2uhr5AmyAfw= =rfHQ -END PGP SIGNATURE- ___ --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: They are? Re: [Asterisk-Users] Now that Nufone is dead...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 23 May 2006, at 16:35, Andrew Kohlsmith wrote: On Tuesday 23 May 2006 10:48, Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? That's news to me; I terminate about 5kmin/month through them, except for about 1 week this month when their carrier dropped them. They are most certainly back up and running. 800 origination is also reportedly working, although I don't have any 800#s from them. It is working, I have a 800-number with them. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEczcKRAx5nvEhZLIRAjAFAKCSXUriFKf0dXeMZJd/CJ12mKknMgCgmI0i rObNUjSDhhXv8dyBbNq+bbY= =/ePN -END PGP SIGNATURE- ___ --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] Now that Nufone is dead...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 They bill you for having the 800 number? I thought they only did that for Michigan DIDs. They only bill my actual call time. jens On 23 May 2006, at 16:54, Tom Vile wrote: Then you are a luck one aren't you. Haven't had my 800 number for over a month now but they still bill you for having the number. Interesting. On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 23 May 2006, at 15:48, Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? Nufone is not dead, works perfectly fine for me. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEcybORAx5nvEhZLIRAnAoAJwJ0Ig4EUdrfw1RhTe8ULxzzq3dQQCfesYc +U9WV0uDc/qD2uhr5AmyAfw= =rfHQ -END PGP SIGNATURE- ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEc0ZKRAx5nvEhZLIRApGTAJ9j1aAK2LpQQVqli2uNrOoxFBL4GQCfTUFr wyuiw+R12uRQkTp0ZGZTEF0= =b0Jk -END PGP SIGNATURE- ___ --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] Now that Nufone is dead...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I see it now on the FAQ, but this must be a new thing. I paid $50 in December 2004 and still have over $39 (yes, I don't use it often). If I remember correctly the 800 DIDs were advertised as free of monthly fees, call fees only. jens On 23 May 2006, at 20:13, Tom Vile wrote: $2.50 p/month for 800 DID. On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 They bill you for having the 800 number? I thought they only did that for Michigan DIDs. They only bill my actual call time. jens On 23 May 2006, at 16:54, Tom Vile wrote: Then you are a luck one aren't you. Haven't had my 800 number for over a month now but they still bill you for having the number. Interesting. On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 23 May 2006, at 15:48, Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? Nufone is not dead, works perfectly fine for me. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEcybORAx5nvEhZLIRAnAoAJwJ0Ig4EUdrfw1RhTe8ULxzzq3dQQCfesYc +U9WV0uDc/qD2uhr5AmyAfw= =rfHQ -END PGP SIGNATURE- ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEc0ZKRAx5nvEhZLIRApGTAJ9j1aAK2LpQQVqli2uNrOoxFBL4GQCfTUFr wyuiw+R12uRQkTp0ZGZTEF0= =b0Jk -END PGP SIGNATURE- ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEc3NcRAx5nvEhZLIRAvRmAKCILQBQw9vKvbccNJ3KG5Tetj7ffwCcCTeu BRgBSM9TKY+BETZ0TlnFWQc= =7Byt -END PGP SIGNATURE- ___ --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] Now that Nufone is dead...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Here's one for all the naysayers: I only sent an email to NuFone accounting to inquire about that $2.50/month fee and they're falling over themselves to not only get all my questions answered but to also helping me getting my account set up in the most economical way for me after their upstream provider problems. Proves me right for sticking with them. jens On 23 May 2006, at 21:41, Jens Vagelpohl wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I see it now on the FAQ, but this must be a new thing. I paid $50 in December 2004 and still have over $39 (yes, I don't use it often). If I remember correctly the 800 DIDs were advertised as free of monthly fees, call fees only. jens On 23 May 2006, at 20:13, Tom Vile wrote: $2.50 p/month for 800 DID. On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 They bill you for having the 800 number? I thought they only did that for Michigan DIDs. They only bill my actual call time. jens On 23 May 2006, at 16:54, Tom Vile wrote: Then you are a luck one aren't you. Haven't had my 800 number for over a month now but they still bill you for having the number. Interesting. On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 23 May 2006, at 15:48, Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? Nufone is not dead, works perfectly fine for me. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEcybORAx5nvEhZLIRAnAoAJwJ0Ig4EUdrfw1RhTe8ULxzzq3dQQCfesYc +U9WV0uDc/qD2uhr5AmyAfw= =rfHQ -END PGP SIGNATURE- ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEc0ZKRAx5nvEhZLIRApGTAJ9j1aAK2LpQQVqli2uNrOoxFBL4GQCfTUFr wyuiw+R12uRQkTp0ZGZTEF0= =b0Jk -END PGP SIGNATURE- ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEc3NcRAx5nvEhZLIRAvRmAKCILQBQw9vKvbccNJ3KG5Tetj7ffwCcCTeu BRgBSM9TKY+BETZ0TlnFWQc= =7Byt -END PGP SIGNATURE- ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEc4i/RAx5nvEhZLIRAhuMAJ4024Ve2xV8Izfb/w6lZfrMhXOpdgCeJVhT Po9tYurKrCRkKAUpNYQ5wXo= =lD7Q -END PGP SIGNATURE- ___ --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: They are? Re: [Asterisk-Users] Now that Nufone is dead...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 23 May 2006, at 23:55, Alexander Lopez wrote: Are the 800 numbers you have new (post-outage) of existing (pre-outage)?? SNIP It is working, I have a 800-number with them. I have a new one right now, but have initiated the porting process to port the old one. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD4DBQFEc5TmRAx5nvEhZLIRAlFlAJieZiUFf1OwWUcjBSAaOwSzdTn4AJ0ahSfG 9vOM5BPH3n1VeESaSDF/fA== =fKq4 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On 21 Apr 2006, at 18:21, Olivier Krief wrote: To benefit from DIVA Server 4BRI fax hardware capabilities, what is the best software combination ? Asterisk and Hylafax ? Shall we then allocate destination numbers and or ports for each of those 2 applications ? And if you want to offer to every user, a unique extension for fax and voice, would it still be possible to forward calls from voice application to fax application (for outgoing faxes, the fax application can use its own ressources) ? I run the combination of Asterisk and Hylafax, and it is easy for me because I have 3 incoming MSNs. Both Asterisk and Hylafax will see all calls, but Asterisk only has two of the three MSNs configured. The fax number is ignored by Asterisk, so Hylafax answers that after a couple rings. This works perfectly fine, but unfortunately I fell into the DIVA V- BRI doesn't do FAX trap, I bought a V-BRI first. It's now sitting on the shelf, unused. The Asterisk server is using the standard BRI card, which I bought off eBay. IMHO the Eicon website should carry more prominent warnings/ explanations about the lack of FAX capabilities for the V-BRI cards. jens ___ --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] Which Mac OSX softphone with IAX2 support?
On 23 Mar 2006, at 23:48, Mike Dent wrote: Hi, which OSX softphone do you use that supports IAX2 protocol with Asterisk? I like LoudHush a lot: http://www.loudhush.ro/ It is a very simple client, but looks great and works well. My only complaint is that the ring tone it generates when you call someone is really annoying. jens ___ --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 Users Mailing List Traffic
On 19 Mar 2006, at 17:47, Ira wrote: At 07:50 AM 03/19/2006, you wrote: Actually, for something like Asterisk, that has so many different aspects, a Forum would be a much better idea. Then, each piece of hardware can have its own category, along with an FAQ. There's lots of Asterisk forums out there already, but weirdly enough all the really good information is in the mailing list. Funny how that seems to be a consistent pattern in the world. Whether or not a forum is a better idea isn't really depending on the subject matter IMHO. Its success or failure depends on what the prospective participants like better. I personally cannot stand forums. That's a place where I have to expend energy to go there and manually click through stuff. If I remember to go there and say up to date, that is. Email comes to me, and is sorted suitably on the server side so there is no clutter. Deleting messages I don't care about is much easier than clicking myself through some thread on a forum. jens ___ --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] Fedora Core 3 or Fedora Core 4? yum update ornot?
On 8 Feb 2006, at 09:43, JP Carballo wrote: Alex Barnes wrote: I think the once it's working, leave it alone advice is very sound indeed :) A similar rule says If it ain't broke, don't fix it. Until you realize some script kiddie has exploited another Apache/ mod_ssl bug and is now remote-controlling your box. There are no hard and fast recipes here. Neither the automatically apply any and all updates nor the build and never look at it again- policies should be applied without taking the specific situation into account. If your box is on the internet you simply cannot forego updates. Period. If your box is completely walled off from the internet you can be lax about it (unless you have to worry about attacks from the inside). The best policy is probably one that is halfway between the two. There are packages you only ever want to update under parental supervision, like kernels. Then there are packages where you want to grab any update you can get ASAP, like Apache, or PHP, or SSH. Yum allows you to express this in its configuration, you can exclude packages from the automatic update. I personally run a nightly script that uses yum to determine if there are updates. I apply them by hand. However, this is only feasible because it runs on just two machines. jens ___ --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] ISDN Eicon Diva Server V-BRI
On 2 Feb 2006, at 18:25, Bartosz Jozwiak wrote: Dear all, I'm planning to buying Eicon Diva Server V-BRI for my asterisk server and run with chan_capi. Is anybody using that card ? Would appreciate any feedback. Card works great with chan_capi-cm from sourceforge. Don't overlook the small print: the V series cards don't do any fax. The standard Diva Server BRI cards do (I assume). The V cards are branded as optimized for voice, but at this point I'm not sure what the advantage really is in a Asterisk system. Had I known about the Fax situation I would have gone for the standard card, which I believe is a little more expensive. jens ___ --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] Interface card for Euro-ISDN (BRI)
On 31 Jan 2006, at 09:12, John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. I'm very happy with an Eicon Diva Server V-BRI that I bought a couple months ago. The only drawback is that it doesn't do any fax traffic apparently. It works with chan_capi-cm from Sourceforge. http://www.eicon.com/worldwide/products/MediaGateways/diva-server- vbri.htm jens ___ --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] Interface card for Euro-ISDN (BRI)
On 31 Jan 2006, at 10:06, Armin Schindler wrote: I'm very happy with an Eicon Diva Server V-BRI that I bought a couple months ago. The only drawback is that it doesn't do any fax traffic apparently. It works with chan_capi-cm from Sourceforge. The 'V' version of that card is for (V)oice. The standard BRI do support Fax/analog Modem and even RTP with codecs and anti-jitter (echo- cancel too). I'm currently working on support for this CAPI-RTP with chan_capi-cm. Yes, I bought it specifically for the Voice optimizations - but my impression was that this was an optimization that would retain other, more basic functions like handling Fax ;) jens ___ --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] Fritz card technology German *
On 17 Jan 2006, at 21:41, Francesco Peeters (Asterisk) wrote: On Tue, January 17, 2006 22:10, Camilo Gonzalez-Cortes said: The Fritz cards was not designed to run on asterisk whereas the following German ISDN cards (http://www.junghanns.net/en/ quadBRI_produkt.html) was designed specially to run on this platform. The only problem with this vendor is the support...It is terrible. They never respond an e-mail Almost any card with the cologne HFC-S chip will work with their drivers + Florz patch, mISDN or vISDN. In my epxerience vISDN gives the best EURO-ISDN support, but it is a very young project, and still misses crucial stuff like echo cancelling... Don't forget the CAPI-based cards. I'm very happy with my Eicon DIVA Server V-BRI and chan_capi from sourceforge. Haven't had any problems or hiccups from day one after creating the initial setup with my German ISDN line. jens ___ --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] SIP/IAX softphones for use in call centre environments
On 5 Jan 2006, at 09:45, Zoa wrote: Have a look at our idefisk softphone. (available for windows, mac and linux). The download links at http://www.asteriskguru.com/tools/ idefisk_beta.php only lead to Windoze versions, how do I get the Maxc version? Thanks! jens ___ --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] IAx/g729 client for MAC
On 27 Nov 2005, at 13:43, Chris Mason (Lists) wrote: Is there a good quality stable (not free) IAX2 client for MAC? I have a client wants to travel and make calls and I want to avoid the SIP blocking that is a problem for travellers. I have heard good things about http://www.loudhush.ro/ But haven't used it (yet). I couldn't get any of the other IAX2 clients to be stable on the MAC. I've been using Loudhush and really like it. jens ___ --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] What is the best Dell Machine for Asterisk?
On 29 Dec 2005, at 01:02, William Boehlke wrote: The 830s are nice but limited because they do RAID on a card and have but one suitable PCI slot. So you can have an interface card or RAID, but not both. That's not true. I just built a system on a Dell 830 with the RAID card. There are three PCI slots in total and one of them fit the Eicon DIVA Server card I'm using. I've been using Dell rackmounts at work for years now and never had any issues. This is the first time I went for a tower server, there is no rack at home... The box isn't entirely noise-free, but compared to the equivalent rackmount models it is very quiet, you could call it pantry-friendly as opposed to living room-friendly. The machine runs CentOS 4.2 (RHEL 4.2 with the VIN numbers scraped off) and Asterisk 1.2.1 with chan_capi. Granted, I'm not a heavy user, but I like the voice quality I'm getting. jens ___ --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] smsq
On 29 Dec 2005, at 15:28, chris songer wrote: has anyone had any luck compiling and installing the smsq.c utility. I went through the tutorial online and found i was getting errors all the way through it. this is the tutorial i was using... http://www.voip-info.org/wiki-Asterisk+cmd+Sms any light on this subject would be greatly appreciated. It is part of Asterisk 1.2.1 (that's what I have here, not sure about earlier versions). jens ___ --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] Eicon DIVA Server V-BRI questions
Hi *, Coming from a very simple and nicely working pure IP setup I'm now trying to converge the IP setup with my real (German) ISDN phone line. I bought a Eicon DIVA Server V-BRI-2 and it is currently connected like this: PSTN - DSL Splitter - NTBA - DIVA The NTBA has two S0-connections, the second one is still hooked up to a T-Com T-Eumex 520 PC which allows me to connect analog devices, that's how my phone and fax works currently. I went to the Eicon website and downloaded the latest version of their driver package divas4linux_EICON, version 8.0beta1. Using their configuration tools, the card is currently configured this way: D-Channel protocol - 1TR6 - Germany Interface mode - TE DID - no D-channel layer 2 activation policy - only by other side Trunk operation mode - Point to Multipoint Upon system start I am getting the green Layer 1 light on the card's back and the following system log messages, which to me looks like the drivers are loading correctly: Eicon DIVA - DIDD table (http://www.melware.net) divadidd: Rel:3.0 Rev:1.13 Build:105-92(local) Eicon DIVA Server driver (http://www.melware.net) divas: Rel:2.0 Rev:1.46 Build: 105-92(local) divas: support for: BRI/PCI PRI/PCI adapters divas: Diva Server BRI-2M 2.0 PCI bus: 0006 fn: insertion. ACPI: PCI interrupt :06:00.0[A] - GSI 11 (level, low) - IRQ 11 divas: Diva Server V-BRI-2 IRQ:11 SerNo:35681 divas: started with major 252 Eicon DIVA - User IDI (http://www.melware.net) diva_idi: Rel:2.0 Rev:1.25 Build: local diva_idi: started with major 251 diva_mtpx: no version for struct_module found: kernel tainted. diva_mtpx: module license 'Eicon Networks' taints kernel. divacapi: Unknown symbol detach_capi_ctr divacapi: Unknown symbol capi_ctr_ready divacapi: Unknown symbol capi_ctr_handle_message divacapi: Unknown symbol attach_capi_ctr CAPI Subsystem Rev 1.1.2.4 Eicon DIVA - CAPI Interface driver (http://www.melware.net) divacapi: Rel:2.0 Rev:1.24 Build: 105-83(local) kcapi: Controller 1: MTPX101 attached kcapi: card 1 MTPX101 ready. kcapi: notify up contr 1 capi20: Rev 1.1.2.3: started up with major 68 (no middleware) --- The problem I am having is that according to the isdn4linux page when calling in the card should recognize the call and note this in /var/ log/messages (like Call from X, ignored). It does not do this at all. Also, if I disconnect the T-Com T-Eumex unit so that the server is the only ISDN unit connected to the NTBA and call in I get a message played back by the phone company that the number is not reachable. At that point the green Layer 1 light on the card turns off. To me this sounds like s severe misconfiguration on my part. Is there anyone on the list who is using a DIVA Server V-BRI card in Germany who could help? After digging through all kinds of websites I am also confused about the relationship between the CAPI drivers included with the Eicon software and BRIStuff. When using chan_capi, do I need BRIStuff and zaptel at all? Thanks for any insights, and a wonderful holiday period!! jens ___ --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] Eicon DIVA Server V-BRI questions
On 26 Dec 2005, at 13:54, Jens Vagelpohl wrote: The problem I am having is that according to the isdn4linux page when calling in the card should recognize the call and note this in /var/log/messages (like Call from X, ignored). It does not do this at all. Also, if I disconnect the T-Com T-Eumex unit so that the server is the only ISDN unit connected to the NTBA and call in I get a message played back by the phone company that the number is not reachable. At that point the green Layer 1 light on the card turns off. The mistake was in the D channel protocol - switching from 1TR6 to EuroISDN allowed me to test the card successfully using the Eicon tools. Asterisk now also shows the call coming in. My second question remains: Do I need BRIStuff? I guess I don't seeing how defining a simple extensions context with just Answer() and Echo() works through chan_capi when I tell Asterisk not to load any of the chan_modem* and chan_zap, and the zaptel module is unloaded... :) jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Eicon DIVA Server V-BRI questions
On 26 Dec 2005, at 15:36, Stefan Tichy wrote: Since you bought a Eicon Diva Server card you have to use chan_capi. IMHO you should use current CVS source from http://sourceforge.net/projects/chan-capi/ (or wait for chan_capi-cm-0.6.2) Thanks Stefan, I downloaded version 0.6.1 and in my extremely limited testing this seemed to work OK. I can switch over to the current CVS HEAD if you think 0.6.1 has issues. Are there any? jens ___ --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] Cron still running after uninstalling asterisk
On 13 Nov 2005, at 13:08, Tim Ashman wrote: Can someone help me here. I installed asterisk briefly just to see if it would install on my suse 9.3 system and now I can get rid of a cron job that goes every minute. I've deleted the asterisk user, looked in all of the cron files I can think of but it is beyond me. Here is the line that keeps trying. The directory referred to doesn't even exist anymore. Nov 13 10:06:01 home /usr/sbin/cron[21829]: (root) CMD (/var/lib/asterisk/agi-bin/run_wakeups) Not sure about SuSE, but on RedHat(ish) systems, cron jobs can be... - defined for specific users (written out to /var/spool/cron) - inside /etc/cron.hourly|daily|weekly - inside /etc/cron.d - written into /etc/crontab directly (an indicator for bad systems administration) jens ___ --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] NAT and SIP.conf update.
I've found a novel work-a-round: I have a server on the Internet in a data centre that maps a real static address to the dynamic IP address of the computer connected via. the ISP. I've got a script that runs on the client ISP connected machine (its running Linux an the script is in the ppp-up.d directory so runs automatically everytime the pc reconnects and gets a new dynamic IP) - this client script (perl) talks to the server daemon (also perl) and then the data centre server re-maps the static IP to the dynamic IP - it redirects (using socat - excellent software!) to redirect the IAX2 port and the RTP ports between the IP addresses so that the normally dynamic IP addressed asterisk server now always has a real live static IP address. This solution is working very well between two remote offices passing calls between the central data server computer. I wonder would this solution help many people? I now couldn't live without it. If other people are interested I could have a little business renting people static mapped IP addresses. Who needs that when there's dyndns and similar free services which are even supported by many routers? I have a dyndns hostname and my router is configured to contact the dyndns site whenever the IP on the public side changes. Works very well for my Asterisk setup at home. jens ___ --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] NAT and SIP.conf update.
On 13 Sep 2005, at 11:18, razza wrote: Jens Wrote: Who needs that when there's dyndns and similar free services which are even supported by many routers? I have a dyndns hostname and my router is configured to contact the dyndns site whenever the IP on the public side changes. Works very well for my Asterisk setup at home. I'm sure if you use a DNS in SIP.CONF for your external IP this is only resolved when loaded? This might be true - for me there's only other Asterisk servers connecting from the outside using IAX, and that works fine. jens ___ --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] zttest
On May 16, 2005, at 14:37, Rich Adamson wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). How do you disable hyper threading (what's the command and where is it placed)? If this is a Linux box, look at the kernel boot arguments in [lilo| grub].conf and append noht, that disables it. My grub.conf on one of my boxes looks like this: title CentOS (2.4.21-27.0.4.ELsmp) root (hd0,0) kernel /vmlinuz-2.4.21-27.0.4.ELsmp ro root=LABEL=/ noht initrd /initrd-2.4.21-27.0.4.ELsmp.img jens ___ 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] zttest
On May 16, 2005, at 19:04, Damian Funnell wrote: ...Jens makes a liar out of me, although I read that the 'noht' switch stops the OS from using H/T but doesn't disable it completely. I make no warranties regarding the accuracy of this information, though. OK, let me rephrase it: After using noht top showed the physical number of CPUs again, not double that. That's the one thing I can confirm! ;) jens ___ 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] Look at that Digium Broadband Modem!
On Mar 26, 2005, at 7:47, Remco Barende wrote: Correct, but I've also seen many reports that replacing the power cube of the IAXy with one that can provide ample power did solve the problems and even resurrected 'dead' IAXY's. You know, since the only power supply that came with the IAXy when I bought it was for US-style outlets and I'm in Germany the first thing I did was throw away the original brick and connected a generic one where you can select voltage and polarity. Hearing about the possibility of power supply problems several times now made me think that the reason I haven't had any of the problems is because of the much-stronger (in terms of available amperage) power brick I use. A few days ago I had the first-ever problem with the IAXy, after probably three months uptime. I picked up the receiver and hear static instead of a dial tone. power-cycling the device fixed it. jens ___ 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] why even use SIP
- proper DHCP and possibility of static IP Never had a problem with mine. I set my DHCP server to hand out a specific IP to the IAXy, too. - a 'reset' button What's the advantage over unplugging the unit and plugging it back in? And my IAXy doesn't work with my european phone (no tone) it's kind of a drag :( My IAXy works perfectly fine with a cheapo Panasonic cordless I bought here in Germany. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: AntiSpam Alert from Rusten McKenzie
On Mar 15, 2005, at 19:04, dean collins wrote: Is there anyway we can get this shit off the asterisk list apart from posting their email address [EMAIL PROTECTED] here for the spambots to pick up? I believe these do not go to the list, but to people who post to the list. You cannot turn it off at the list level. IMNSHO this approach where you make people go through hoops to get mail to you is utterly unfriendly in nature. Those people should look at spam filters instead of making it inconvenient for everyone to send them mail. jens ___ 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] Apple links Asterisk
On Mar 10, 2005, at 6:31, Matthew Boehm wrote: From macintouch.com: Apple is distributing an open-source Asterisk install package for Mac OS X: I suppose they get a little overexcited. Apple isn't distributing anything, they just link to a third party that made a ready-to-install package. That link has been up since August 2004, and the Asterisk version it uses is CVS 10-28-03... yikes :) I might be interesting to build from a recent source and extract the extra pieces they advertise out of that installer package. jens ___ 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] Limit the call recording when pressing *1
On Feb 27, 2005, at 8:11, Joseph wrote: Though, I'm not sure in what value is the time expressed. When I input (6000:5999:1) as soon as I pickup the phone the time was announced I have only 5sec. left As the source you pasted in your original post clearly states, the value is in ms. That's not minutes, that's milliseconds. jens ___ 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] Wierd asterisk-perl compilation problem
On Feb 26, 2005, at 18:52, mattf wrote: A good rule of thumb for heavy perl users is to not use Fedora/RedHat. Or at least not use rpms or the preinstalled perl on the OS. RedHat has done a lot to screw up how perl works in the last several versions and there are a lot of angry perl developers that have just given up on the distro altogether. Funny thing is, this is true for Python as well. No one who cares about the things they run on it should *ever* tie themselves to the distribution's package. Compile your own is the standard recommendation and solution. jens ___ 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] Getting speex to work
On Feb 24, 2005, at 1:35, Jonathan Lin wrote: I can see libspeex.so.1 in /usr/local/lib and it's symbolic linked to libspeex.so.1.2.0 so the only thing I can think of is the permission. I changed the permission to 777 for libspeex.so.1.2.0 just for testing but it's still crashing. Has anyone encounter this problem or maybe point me in the right direction for debugging this? Probably because it's in /usr/local/lib, which might not be recognized as a valid library path. On RH-based systems I would add /usr/local/lib to /etc/ld.so.conf and then run ldconfig. jens ___ 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] bridging iaxtel calls to PSTN
On Feb 22, 2005, at 21:15, Brian Capouch wrote: That's for starters. I'm sure others will chime in with other evils beyond these. HTML mail is a favorite tool for virus writers and spammers because it's so easy to hide nasty payloads and all those helpful garbage email clients out there love to fetch and render whatever some unknown sender tells them to... jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Call Monitoring
BTW, I did need to suid the zttool-cli command to root, as the normal BB user doesn't have the needed permissions. I haven't looked into this, but if anyone has a suggestion on a better way to do this, feel free to let me know. It's called sudo jens ___ 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] bridging iaxtel calls to PSTN
On Feb 21, 2005, at 23:36, [EMAIL PROTECTED] wrote: Hello, actually I did, but nobody responded to that. Maybe people would look at it if you stopped sending HTML mail. jens ___ 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] Speech Recognition
On Feb 12, 2005, at 17:58, Roy Sigurd Karlsbakk wrote: Does anyone know of a speech recognition module (like say yes or no, or numbers) I guess due to the complexity of speech recognition it might just be found in commercial applications or am I wrong like always? What's wrong with the old and non-fancy IVR? Voice recognition menus only piss people off. If you're setting up a call center where you want as many as possible of the customers to ABANDON their calls, go on... How true that is... faced with customer-unfriendly service like that (especially when they don't offer a choice to get a human at all) I start hitting keys like 0 or # or * until something happens... jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE:mandrake linux install of zaptel
On Feb 11, 2005, at 16:28, [EMAIL PROTECTED] wrote: Extreme N00b, I am getting the error message a target does not exist when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. I think everyone would appreciate if... - you wrote a new mail instead of highjacking an existing thread by answering it and replacing the subject line - you would not keep 5 miles of completely unrelated stuff in your email message - you could provide a better problem description that includes specific error messages and message stacks. Thanks! jens ___ 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@home scary log
On Feb 10, 2005, at 17:12, denon wrote: Why would you even want SSH exposed to the world? In fact, why expose it to anything but your local admin console, or *maybe* a vpn tunnel server if absolutely necessary? SSH is perfectly fine, but the first thing I do is disallow any *password-based* access. Only SSH key access is allowed, ever. jens ___ 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] Problems DIALING to IAXTEL
On Feb 6, 2005, at 8:18, Gonzalo Gasca wrote: Do the ECHO TEST dialing to 17002353660 Make 4 calls first one completed succesfully, second FAILED, third and fourth were succesful. For the second call i got this ERROR: -- Hungup 'IAX2[69.73.19.178:4569]/1' == No one is available to answer at this time This happens randomly when I dial to IAX (ie Digium numbers) not only this time It's busy. That's all. iaxtel is a *free* service, there cannot be any guarantees that all calls are successful. If you can't live with that you need to spend money and buy from a commercial provider. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: load balancing 20 asterisk servers
On Feb 3, 2005, at 17:08, Jon Bebeau wrote: Mark, I've been following this thread with some interest as we're gearing up for load/failover processing. Can you elaborate on the garp and IP takeover process, like what software packages do that in Linux or point me to a site for more info? http://www.linux-ha.org/ jens ___ 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] IAXy Hung, Power-cycle Required
On Feb 3, 2005, at 17:37, Adams, Gavin-ML wrote: Has anyone had good success with the IAXy? I've tried everything including PAT on the IAX2 port to the IAXy device to no avail (using the alternate server parameter). I guess a call to Digium is in order! Works for me[TM], without fail jens ___ 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] list administrator.....???
On Feb 2, 2005, at 2:50, Greg Hill wrote: ..so can anybody confirm the guess? If the first n-1 digests of the day are roughly the same size, that might support the theory. Yes, that's how Mailman works. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: load balancing 20 asterisk servers
On Feb 3, 2005, at 0:03, Miguel Ruiz Velasco Sobrino wrote: --- [EMAIL PROTECTED] wrote: The DNS approach does not handle single or multiple system failures, only very elementary load balancing over a lengthy period of time. Are you shure of that? I'm aware that the load criteria is trickier, but very possible. Operating systems and probably a lot of devices *cache* the results of DNS lookups. That means removing A records won't do any good. Short story: No matter what network service is being balanced, if you want to guard against failure and against customers noticing that failure use a real load balancing solution, DDNS is not suitable. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: load balancing 20 asterisk servers
On Feb 3, 2005, at 4:20, Matthew Boehm wrote: I'm trying to stay away from a software based load balancer cause what happens if that server fails? Its far less likely for a piece of dedicated hardware to fail than an actual computer. There are useful things like heartbeat which can transparently fail over from one machine to the next. They even take the IP address of the failed machine. A professional setup would have redundancy built in that way. I have run extremely busy load balancers that way and in the failover case everything is back to normal within seconds. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Messaging with * and eyeBeam
On Feb 1, 2005, at 17:55, Ferguson, Michael wrote: -Original Message- From: Ferguson, Michael Sent: Tuesday, February 01, 2005 11:35 AM To: 'asterisk-users@lists.digium.com' Subject: Messaging with * and eyeBeam G'Day All, snip Repeating that message over and over won't get you any more responses. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk
On Feb 1, 2005, at 18:13, Adams, Gavin-ML wrote: snip How many more empty test messages are we going to see from you..? jens ___ 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] problem in compiling asterisk addon
On Jan 29, 2005, at 13:19, Kamran Ahmad wrote: now it is giving another error - [EMAIL PROTECTED] asterisk-addons-1.0.1]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:31:25: mysql/mysql.h: No such file or directory You don't have the MySQL devel package installed. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to use ASTCC with SIP ??
On Jan 29, 2005, at 16:15, Daniel Eboa wrote: image.tiff image002.jpg Hi Daniel, Would it be possible for you to turn off attaching two image files as signature replacements to each of your email and maybe use a text signature instead? Thanks! jens ___ 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] /usr/bin/ld: cannot find -lidn
On Jan 27, 2005, at 15:12, Matt Schulte wrote: Bueller? Is this a lib of some kind? Google and lists bring up nada, this is from ast cvs head latest on Fedora Core 3. /usr/bin/ld: cannot find -lidn collect2: ld returned 1 exit status make[1]: *** [app_curl.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 On my Apple Cube that I use for Asterisk, yum info libidn shows this: Name : libidn Arch : ppc Version: 0.5.4 Release: 1 Size : 569.34 kB Group : System/Libraries Repo : Yellow Dog Linux 4.0 Base Summary: Internationalized Domain Name support library Description: GNU Libidn is an implementation of the Stringprep, Punycode and IDNA specifications defined by the IETF Internationalized Domain Names (IDN) working group, used for internationalized domain names. So you're probably missing the libidn and libidn-devel packages. jens ___ 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] /usr/bin/ld: cannot find -lidn
On Jan 27, 2005, at 16:02, Geoffrey S. Mendelson wrote: On Thu, Jan 27, 2005 at 03:37:10PM +0100, Jens Vagelpohl wrote: On my Apple Cube that I use for Asterisk, yum info libidn shows this: This answers a question I had but did not think would be answered yes. Which cube are you using? Is a G3 300 (old world) minitower fast enough for a small network? I'm only using it for home use. Can't make any judgment call on your situation. jens ___ 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] Data calls with Asterisk
On Jan 24, 2005, at 1:50, Karim Mardhani wrote: I have about 10 remote locations which are collecting some data. I would like to upload that data every night. All remote locations have 56K modem. I was wondering can Asterisk be used to receive this data? Basically I will have an asterisk with 1 FXO card and have it receive data calls. Can asterisk receive data calls? Why use asterisk for that if you can simply plug a modem into the receiving computer and use mechansisms that are *made* for that purpose, such as PPP? jens --- Jens Vagelpohl [EMAIL PROTECTED] Software Engineer +49-(0)441-36 18 14 38 Zetwork GmbHhttp://www.zetwork.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] VoIP Providers and Backbone Servers
On Jan 24, 2005, at 1:51, [EMAIL PROTECTED] wrote: Additionally, these small beginings enable people like myself to learn the industry quickly and get involved. It also allow us to learn about the Astrisk PBX system as well as the multitude of hardware and software that comprise this exciting field. You need to do what you have fun doing - anything else isn't worth doing. As long as you don't overrepresent yourself and/or customers end up being guinea pigs because your learning process has not proceeded far enough you have every right to work on that idea and make it happen. Even if you don't offer some flashy new feature others don't. I wish you good luck. jens --- Jens Vagelpohl [EMAIL PROTECTED] Zetwork GmbHhttp://www.zetwork.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] Autio cut off at beginning of call
On Jan 24, 2005, at 11:27, Senad Jordanovic wrote: Check the load on your server(s). I have the same problem with calls to and from NuFone. It's probably not load-related because the load is non-existent on that box. It runs nothing but Asterisk with a very simple network-only config where no telephony hardware is used. The only thing connected to it is an IAXy with a cordless hanging off it. jens P.S.: Am I the only happy IAXy user out there or what? I love that thing. Never any trouble. ;) --- Jens Vagelpohl [EMAIL PROTECTED] Zetwork GmbHhttp://www.zetwork.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk monitoring with Nagios and IAX
On Jan 19, 2005, at 10:09, Florian Lefeuvre wrote: Hi, What do you want to check exacly? that * is still alive? you want to know the number of concurrent call? The only think I want to find out is if Asterisk is still alive and requests coming in via IAX2 are answered. Just some kind of simple ping (Hello, anyone there) that produces a simple pong (Yes, I'm alive). jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk monitoring with Nagios and IAX (Roy Sigurd Karlsbakk)
it's there already, on http://karlsbakk.net/asterisk/ and under new plugins on http://sourceforge.net/projects/nagiosplug/ Yes, I've looked at your plugin. However, I'm trying to come up with a much simpler setup that does not require access to the manager interface and that does not require any nagios scripts running on the Asterisk box. I want to check upness from the remote Nagios box by simply issuing some kind of IAX ping. I'm going to look into using a network traffic analyzer to capture such a packet and just use that Windoze iaxping binary to generate it. I had hoped I would not need to go that far ;) jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk monitoring with Nagios and IAX (Roy Sigurd Karlsbakk)
I'm going to look into using a network traffic analyzer to capture such a packet and just use that Windoze iaxping binary to generate it. I had hoped I would not need to go that far ;) There: https://sourceforge.net/tracker/index.php? func=detailaid=746083group_id=29880atid=541465 Added IAX ping :) You = Da Bomb ;) I'll play with it a little this afternoon. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk monitoring with Nagios and IAX (Roy Sigurd Karlsbakk)
There: https://sourceforge.net/tracker/index.php? func=detailaid=746083group_id=29880atid=541465 Added IAX ping :) Improvement suggestion: The while loop that checks for an IAX answer currently runs as while (1), so it always runs until the timeout has been reached. I replaced the while (1) with while ($iax_answer == 0) to make it break out of the loop immediately if an answer has come. Works nice :) jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk monitoring with Nagios and IAX
Hi *, Does anyone have a lead on a Nagios plugin that speaks IAX or a small app to do so? I'm trying to set up remote monitoring for my Asterisk server and only IAX2 traffic is allowed through the firewall. Simply using check_udp to port 4569 yields no usable answer and Asterisk complains about receiving a midget packet or something like that :) jens ___ 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 monitoring with Nagios and IAX
an application available called iaxping with would send a set of well formed iax packets and wait for the response. Unfortunately that application was written in visual basic, and no source code was distributed. A skilled coder could probably use some of the required functions from iaxclient or libiax2 and create a similar function in C. I've googled around and tried a few things. The ideal solution would involve no library dependencies to make maintenance easier. The specific problem I have is that I don't know what a IAX packet contains which elicits a measurable response from Asterisk. I have looked at the sources but it's very convoluted and the last time I touched any C was 7 years ago... So if there was a way to synthesize this IAX PING or POKE packet it shouldn't be hard to just package that into a script that handles the connection establishment, sends it, and listens for a response. jens --- Jens Vagelpohl [EMAIL PROTECTED] Software Engineer +49-(0)441-36 18 14 38 Zetwork GmbHhttp://www.zetwork.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] pattern matching problem
On Jan 17, 2005, at 7:29, Joseph wrote: How do I solve the problem with between patterns: _1800 _1NXX I would like all numbers 1800, 1877 etc to go through iaxtel but all other numbers 1xxx via voipjet When you combine these contexts, e.g. when you include them in your default context, you need to make sure that the more specific expression (in this case the iaxtel expression) appears *before* the less specific expression (outgoing-voipjet). First match wins. jens --- Jens Vagelpohl [EMAIL PROTECTED] Software Engineer +49-(0)441-36 18 14 38 Zetwork GmbHhttp://www.zetwork.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] Teleconferencing?
On Jan 13, 2005, at 17:12, Matt Burleigh wrote: I am just now investigating Asterisk. Can Asterisk provide 6-10 party teleconferencing when configured properly? Yes Matt, it can ;) P.S.: Ask Andrew, it's running at ZC --- Jens Vagelpohl [EMAIL PROTECTED] Software Engineer +49-(0)441-36 18 14 38 Zetwork GmbHhttp://www.zetwork.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] Connecting a Home based worker with An Iaxy
On Jan 10, 2005, at 16:56, John Middleton wrote: Hi, If I need to connect a home based user to an Asterisk server, how does the above work? Is it (after being configured/provisioned) plug and play? Anyone done this got any comments Yes it is plug and play. Here at home I have set my DHCP server to hand out a specific IP to the IAXy as well, so configuration becomes even easier. Plug it in, wait a little for it to get its IP, and then use the iaxyprov utility to configure it. I followed the PDF you can download from digium.com. I've been very happy with it and haven't seen any of the issues other people reported (no ring, losing IP, etc). jens ___ 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] Clipping on outbound calls via SIP/IAX
On Jan 2, 2005, at 21:16, Reid Forrest wrote: I'm hoping someone can help me with a problem I've been having for a while now. I've googled and wiki'd to no avail. Whenever I place an outbound call from * to a PSTN through a SIP or IAX provider (e.g. Voicepulse or Broadvoice), the first 1/2 to 2 seconds of the remote call are clipped (muted). For example, if I call a remote voicemail system that usually answers with Nortel Call Pilot, Mailbox? I might get ilot, Mailbox?. Everything works fine if I dial an internal extension or through the PSTN. Is this just something I'm going to have to live with if using an Internet-based termination provider? I'm using Asterisk 1.0.3 and have tested on different systems, different providers, different phones, etc. I have the same symptom, dialing from a phone hanging off a iaxy that talks to my * and then outbound through NuFone. Where I expect to hear Thank you for calling foo... I get calling foo... only. jens --- Jens Vagelpohl [EMAIL PROTECTED] Software Engineer +49-(0)441-36 18 14 38 Zetwork GmbHhttp://www.zetwork.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
