Re: [asterisk-users] Asterisknow with video and X-Lite not quite working
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Benedikt Franz wrote: Hello everyone, I have previously asked this question on the asterisk-video list, but I got directed here. I have a setup consisting of asterisknow beta4 (not sure if that is crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the local network. My computer has a USB-Camera installed, and now I would like to do some video calling with it, at least, so that the other user can see me. When I make a call and then click 'Start' (sending video) in the X-Lite client, nothing seems to happen on the other side, but here it says that a video transmission has begun. According to 'sip show codecs', both the h.263 and h.263p codec are supported, and those are also set on either X-Lite clients. I have enabled 'canreinvite' for both users as well, but still the other user can not see me. I can, however, see the cameras view on my computer, so that seems all properly set up. Could anyone help me sort this out? Do you have: videosupport=yes in the general section of sip.conf? - -- Cheers, Matt Riddell Director ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF96K0S6d5vy0jeVcRAn2KAJ9Mzb3daRKGePBOxWZDXKadylQofQCghzK7 NElGyp/6yB6P1kfvdxNJ2aY= =NZd+ -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: what happened to asterisk wiki???
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 JR Richardson wrote: A friend of mine was on the site yesterday, late morning, when he refreshed his screen, a banner came across the web page VOIP SUCKS and then the site was no longer available. I'm pretty sure the site was compromised by some hacker trying to prove a point or make a statement. Not to throw stink on anyone or group, but maybe it was someone from a competing open source VoIP project or one of the Big Iron VoIP System Manufacturers. Probably just some cracker with too much time on their hands. I feel like someone shot my dog, please get the site back up as soon as possible. There was a post about a security vulnerability in wiki on bugtraq a couple of days ago, but it looked more like someone had figured out how to edit pages (pointless considering a wiki is open anyway). - -- Cheers, Matt Riddell Director ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67 NV4lAmOkaISR79fBTjajGw8= =u7sc -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] FAX using T38
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Underwood wrote: Andrew Joakimsen wrote: I'll consider the offer if it includes your code being included with Asterisk. On 3/3/07, Steve Underwood [EMAIL PROTECTED] wrote: I'll do it for 30% less than they quote. :-) So, you want a discounted price for something that offers more than Attractel will offer you? Do you understand how negotiation works? :-) Steve Hey, Would you be willing to name a price to get Asterisk to do disclaimed T.38 gatewaying (ooh that word can't be right)? I for one would be keen to help run a donation thermometer thingy (I.E. bounty). - -- Cheers, Matt Riddell Director ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF9L9US6d5vy0jeVcRAi+iAJ9j5ThvxOstdjD6xJhsB5RVE4fmjwCeIT4s 77dyJtBQkm163D+AIMXAmP8= =1UXJ -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] Paid support offered
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike Lynchfield wrote: We have decided to allow our tech's to do support for non-clients of voicemeup.com This should normally be kept on the Asterisk-Biz list This list is for Non-Commercial Discussion - -- Cheers, Matt Riddell Director ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5lbFS6d5vy0jeVcRAjjYAJsGiFxYSpbrW3Pd3kUgnZPKMnXeDwCfbO2g 5CCyhrBlD7fx9oQLgdyzkD4= =6tN/ -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] FAX using T38
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andrew Joakimsen wrote: OpenPBX.org has better support, due to license issues and politial bullshit I don't see Asterisk getting T.38 support that isnt a joke (codec pass-thru?? LOL) for a long time. OpenPBX should have a stable release within the month, if I am not mistaken they have a Release Candiate #2 right now Does OpenPBX do a T.38 gateway then? - -- Cheers, Matt Riddell Director ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5leIS6d5vy0jeVcRAijAAKCGTqF1iv/AxZ1H2znpgVlzcsLAwACfQzbJ eJaY9KJ5QXrCRH6a6cnRRrM= =Owpm -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] Call connected, cannot hear or speak - $20 for fix
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Supa wrote: I using my provdier like so SIP/Telasip-gw4/5198843344 when bridging calls. All my local extensions work, so does disa and the like Did you get this going? - -- Cheers, Matt Riddell Director ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5ljXS6d5vy0jeVcRAtiuAJ9m5LOTjFDiPdm+Ux3Ic6nXAPRcaACcDHjC J5Gdt8Rc/BDfi33U8Bku85A= =A2KZ -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] MusicOnHold Files
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Yuan LIU wrote: One item in my todo list is to make better sound quality whenever end point supports it. Wide-band codec's can already produce better sound than toll. So why do we still need to convert to 8 bit? Should be 16 bit, 8Khz, not 8 bit. However, your point stands, that it should be able to use 44.1Khz instead of 8Khz. The problem is, there is still quite a bit of work to support larger sample rates without simply doing passthrough. If I remember correctly, someone did have a patch and a bit of work done for 1.2. Maybe this made it into 1.4? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWeVS6d5vy0jeVcRAqx0AJ41+MMCToCDRvTUrJBipwKoyqdj6QCfeEuG 3INcPOyOmwAbBoNJMNQ8ku4= =Nm2h -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] answer machine detection
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Julian Lyndon-Smith wrote: Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Hi, I'm using it in New York, and we seem to be having good success (on this particular provider) with: [general] initial_silence = 3700 greeting = 2500 after_greeting_silence = 1200 total_analysis_time = 6000 min_word_length = 100 between_words_silence = 50 maximum_number_of_words = 4 silence_threshold = 860 Disclaimer: just use this as a starting point, go into the console with debug and verbose up, and make sure that for every word you speak, it recognises a word, then try again with cellphones instead of landlines. Remember that you're not going to get 100%, some answer machine messages may have been recorded quietly etc. You'll need to also make sure that the upstream provider doesn't answer the call and then provide ringing (as the stanaphone DID we were testing did). Report back how you go on this, maybe we should start a wiki page with settings in different places. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWTGS6d5vy0jeVcRAreVAJ0fyym4B/6vkDuesexxYNlTgt3RBgCcCL7W gTs3lOT7W406rmqhmxxqaqA= =3HcB -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] no unicall on 1.4
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Eric ManxPower Wieling wrote: Anton Krall wrote: This is exactly one of the things that Steve and I discussed a bit ago... when did asterisk turn from an open source project with very good developers into a business that only focuses in $$$? I imagine that happened around the time they sold their soul to the venture capitalists. 8-) Oddly, I download and install Asterisk for free all the time. Oh, you must be using the warez version then. 31337 then aren't you! :D - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWJbS6d5vy0jeVcRArCpAJ9nJUq1NHzN/X8DrCMe7yB8LtNXkwCcCRfj 2KojUWrXmmJ/x55GMwvYZoI= =tUpw -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] Call connected, cannot hear or speak - $20 for fix
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 zero massive wrote: I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one that is able to get this going I would be will to give $20 to (via paypal) Does it work when you call from one of the phones to the other? Say in the script you are trying to connect User01 with User02. If you make a normal call between these users, is the audio passed? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWNCS6d5vy0jeVcRAoORAJ4zd5etQcQqntLSdxTWaCzqMwF78ACfW4Jz to2/ubpJXIU+7mQSVfvIIIM= =e0/A -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] Call connected, cannot hear or speak - $20 for fix
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Supa wrote: Yes, it seems to fail when both extensions are external It seems more like a NAT problem than a script problem. Are the phones both connected via SIP? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoXZVS6d5vy0jeVcRAua7AJ9qOZuqlSEeqfWk5cG2v8nt7Buv5gCfVZSm to9odkqByRjDtwaKHogijEk= =g4bZ -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] Odd hangup problem TDM400P
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 joe a. wrote: On an Asterisk 1.2.12.1 system using a TDM400P with two FXO and two FXS modules, servicing two POTS lines: When dialing a number, such as a bank, or pharmacy, where it is required to enter a long series of numbers via the phone's keypad, an unexpected hangup occurs. Do you have something defined to disconnect the call in features.conf maybe? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFl3CuS6d5vy0jeVcRAlgUAJ0TPhZgdRcTH1fDM75HhaICp/ixlgCbB7sC i9/45QmHBlYpCwgha7/NlOU= =DBZ6 -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: FW: [asterisk-users] cdr_addon_mysql.so did not register itselfduringload
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Savoy, Kevin - Williston, ND wrote: Ok so something is missing. I get the below for those two lines. checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... no I even installed the mysql-devel as Bradley Watkins suggested and still it says no. What do I need to make that say yes? Try a make distclean in the addons directory before doing a ./configure. mysql-devel is definitely what you need. Run ldconfig maybe? You shouldn't need to though. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFlLv9S6d5vy0jeVcRAnFwAJ0YER87bSGBWIpQVWt8zRJQnHhq0wCfV1Z8 Vu+U0ejq3sHAfAlDcBilXUM= =0qw+ -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] Happy X-mas
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mitcheloc wrote: Ditto, Happy Holidays everyone! While its hardly on topic, I thought I'd post the following link: http://www.elfyourself.com/?userid=5bfbd9093534410339679acG06122316 Happy holidays everyone! - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFjviBS6d5vy0jeVcRAgCXAJ43fGzXaiZQC+3HDeiIin1UXCqLdACghd5p 2UIJTSjNrLIb+6sOjTHJMfU= =D4Pb -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] Determining invalid extensions.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Phil Finkler wrote: I used _XX. Since it was used in the examples I got from voicepulse. Maybe I can modify it so it's standardized by using 's'. Any idea why they'd use something like that for incoming calls? Are you sure 600 would match _XX.? I thought _XX. Was just two digits. The fullstop will match one or more digits: http://www.voip-info.org/wiki-Asterisk+Dialplan+Patterns - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFjdOvS6d5vy0jeVcRAmFHAJ0XR/VxhpfbM8Kc5ph915JLxCee5gCgjzit 1AL8DrT2EOPpSlVJiHQhpx0= =Nost -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] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: 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. Why not just use different contexts for each company? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE416ZS6d5vy0jeVcRAkkJAJ9ePGEV4H5GNOljhx+syWb42IdoRACfcSet 6dTJAdgseqkUk63mGTOONik= =2M0q -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] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. Surely all the more reason to do it with contexts than instances. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE422LS6d5vy0jeVcRAhQQAJ9XLDlNHe2Xv7oBA568nvaPbnKI1wCeM+t4 5geXNT+XaPj1gSxdSROQKYE= =AZoL -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] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Because Asterisk wasn't designed with carrier class features in mind. It was designed for a single enterprise. The dialplan, and config files, start to get very very complicated after you add more than a few companies. Combine that with having to have multiple extensions for a single function (our Queues are accessed by a regular extension but then have to dial another 'virtual' extension so that DUNDi can work out the 'primary' server for a queue) and so on. Anyway, it's becoming unmanagable. So write better management software, that's what we've and many others have done. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE43DHS6d5vy0jeVcRAnEyAJ9yNLv+vDF2esy4S6Hik8C46POiDQCeKd5X 6BND4aXxRw5nxifVC1oQM6U= =pfPk -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] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: 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. Our first customer has 15 contexts (right now) in extensions.conf (we've broken each company into a separate files included from extensions.conf and sip.conf for some manageability). At several hundred companies, that's several thousand contexts. We have three Asterisk boxes, and can add more, but the config is (almost) idential between them for redundancy, and this means that each Asterisk box has to have a dialplan configured for all companies. And so you're thinking it would be better to run several hundred Asterisk instances?! Good luck. I think your project would work a lot better if you worked like this: 1) Get requirements 2) Map features and limitations of products 3) Write PseudoCode 4) Work out ways to load test your ideas 5) Write real code 6) Load test again with real code Hint: Layer your system so that each component is not doing too much Hint #2: Read: http://www.astricon.net/files/David_Zimmer_EUR06.pdf - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE43N2S6d5vy0jeVcRAj0OAJ4vgp3aMeBiEsVsU+zqhyouu8CPlgCffPAv 0SccdLfefS8GUtkxZpIMpU4= =ciMO -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] Is anybody moderating this list?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Zeeshan Zakaria wrote: Hi, It doesn't seem that anybody is moderating digium's mailing lists, thats why some uncivilized people with no manners to talk keep making this list dirty. Recently I've noticed increase in irresponsibly typed and rudly answered messages. If there are moderators here, they should stop it and kick these people out of these mailing lists. If someone has broken the lists policy post a mail to the list owner. Some people have been banned for posting commercial mails to the list or for spamming, but not being friendly is hardly something to moderate for. If you don't like someone's response, don't read it. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE4Z4AS6d5vy0jeVcRAsynAJ9DkkDck1MdNcDqhnAFQAQ17eVL6wCfRnj/ lINZG4VpdJcJ/DDFjqIrzoM= =7iDZ -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] Run As User Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Forrest Beck wrote: Does anyone have a listing on file/directories that asterisk needs ownership of to run as a user other than root? I know about the major items --- /etc/asterisk, /var/spool/asterisk/, /var/lib/asterisk, etc... Anyone have a script to fix all the directories? vi /etc/asterisk/asterisk.conf maybe? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE4Z4fS6d5vy0jeVcRAt8VAJ9nOO2tYL3nGavDzs8GJHuyKxIn9gCeMx3V BoTXDsieNyGL7p3nmEuoHBU= =ci5J -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] AGI doesn't execute PHP5 script
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Stefan-Michael. Guenther (in-put GbR) wrote: Hi, Am Dienstag, 8. August 2006 13:41 schrieb Matt Riddell (NZ): Stefan-Michael. Guenther (in-put GbR) wrote: Hi, I'm trying to start a PHP5 script via the AGI Interface. The asterisk version is Asterisk 1.2.5-BRIstuffed-0.3.0-PRE-1k and I followed the instructions on http://www.voip-info.org/tiki-print.php?page=Asterisk+AGI+php The problem is, as you can see from the output in the CLI, that Asterisk claims that it executes the script, but nothing happens. It doesn't create the file /tmp/asterisk and it doesn't send an email. When I execute the script manually on the command line, it is executes without an error, the file is there and the email, too. ^^^ Try running it from the command line and see what happens I guess you meant the test.php script, right? Executing php5 scripts on the command line isn't a problem at all, only when they are started through AGI. This particular script also? Are you using AGI DEBUG in console? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE2XwPS6d5vy0jeVcRAlCtAJoDPCIIgX4XxVELnoQYEmvc1l+oLwCcDioH 7AyLQZcKjqrJMBxqNiM4qI8= =tgJm -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] AGI doesn't execute PHP5 script [SOLVED]
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Stefan-Michael. Guenther (in-put GbR) wrote: Yeah. That response is usually when things are not happening properly. Matt, the ouput hasn't changed, although ist script is executed properly. Why do you thing that this output shows a failure? :) Usually my scripts have hundreds of lines of debug statements, so when I see nothing come back, I'm always a little concerned! :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE2cYRS6d5vy0jeVcRAt8WAJ9W15a0rslNlGx8YDXl13dwbFMI0gCfWX8c 58YUsSa6/3Bpr/yyd/VYcps= =r7Pv -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] Ignoring the # key on a call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 patrick wrote: I'm using Asterisk 1.2.7.1, and if I press the # key when I'm on a call, I get a prompt to transfer the call. This, of course, interferes with any IVR system I'm using, as many systems will ask me to enter a number, then press the # key. Is there any way I can get Asterisk to ignore this key when I'm on a call? Have a look inside features.conf, where you can change the hash transfer (or pound) to ## or whatever you'd like. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE2j5gS6d5vy0jeVcRAsORAJwKfICCPAfBK5UQtPnd+un44wHFZgCfbZJh GEByq1FUGXXl6xw+CG3NLMo= =wrcV -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] Fwd: * and GTalk testing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 David Freeman wrote: I don't think this is realted, but I had to eliminate zttranscode from being compiled as it kept killing * after a few seconds. Since I'm not using any pstn (or hardware for tha tmatter, ztdummy) I don't think I need it? Like I said, still learning here, but I want to make sure I cover all the oddities I had to do. Try upgrading. This shouldn't have caused the problem. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE2EsAS6d5vy0jeVcRAqkWAKCPMuYDsSuC94BaRJYDLDoZkrrJ5ACfZJ1f Zd/RLqZzuTEWw6JHKN40Gko= =e6/7 -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] AGI doesn't execute PHP5 script
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Stefan-Michael. Guenther (in-put GbR) wrote: Hi, I'm trying to start a PHP5 script via the AGI Interface. The asterisk version is Asterisk 1.2.5-BRIstuffed-0.3.0-PRE-1k and I followed the instructions on http://www.voip-info.org/tiki-print.php?page=Asterisk+AGI+php The problem is, as you can see from the output in the CLI, that Asterisk claims that it executes the script, but nothing happens. It doesn't create the file /tmp/asterisk and it doesn't send an email. When I execute the script manually on the command line, it is execute without an error, the file is there and the email, too. ## ;extensions.conf ; [guenther] exten = 111,1,Answer() exten = 111,2,AGI(test.php) exten = 111,3,Hangup ## ls -l /var/lib/asterisk/agi-bin/test.php -rwxr-xr-x 1 asterisk root 340 Aug 8 10:07 test.php ## cat /var/lib/asterisk/agi-bin/test.php #! /usr/bin/php5 ?php ob_implicit_flush(false); set_time_limit(6); error_reporting(0); Try running it from the command line and see what happens - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE2Hh3S6d5vy0jeVcRAkf7AJ0bbLWCZXQ/ismZJxG2l2vfMCktiQCfeshM Sk03IFfNGOdKGM9CZvsrldE= =CNyN -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] Fwd: * and GTalk testing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 David Freeman wrote: Got blasted for sending this to the wrong list, anybody using GTalk extensions? :) Yes and no, I've written a bot that responds to messages, but have the same problem as you with audio. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE122FS6d5vy0jeVcRAmaXAJ419kfuhKwgsmBIQdx9TcNI04rfqwCfVD6t Z+iVW73SSMuxnvjAhWagY5c= =oLWU -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
[asterisk-users] [Solution] Call Asterisk from GoogleTalk and have it tell you the status of your IAX2 links.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
After following the instructions for getting GoogleTalk to talk with
Asterisk using Matt O'Gorman's code and Farruk Ahmed's howTo (
http://www.sineapps.com/news.php?rssid=1407 ) I have been using the
Jabber and Jingle additions in the SVN Trunk version of Asterisk and
have come up with a cool idea.
Basically, you can call your Asterisk server from GoogleTalk, and it can
send you the status of a few of your other servers.
All you have to do is set up an account in jingle.conf, i.e.:
[voipproducts]
[EMAIL PROTECTED]
disallow=all
allow=ulaw
context=checkStatus
connection=asterisk
Where [EMAIL PROTECTED] is my googleTalk address (the one I will be
calling from).
Then set up a few simple accounts in iax.conf, i.e.:
[fv2fv1]
type=friend
host=freevoip.gedameurope.com
secret=mysecret
qualify=yes
where freevoip.gedameurope.com is the host I want to monitor.
Then add a bit of extensions.conf:
[checkStatus]
exten = s,1,NoOp(Incoming Call from [EMAIL PROTECTED])
exten =
s,2,JABBERSend(asterisk,[EMAIL PROTECTED],freevoip.gedameurope.com
status is ${IAXPEER(fv2fv1:status)})
Then reload.
Assuming you have set up Asterisk to appear as a google talk contact,
all you then have to do is call Asterisk (in my case using my
[EMAIL PROTECTED] address), it will send the call to checkStatus
and then reply with the IAX2 ping time in a message!
Cool yeah? Oh yeah, and you could add as many of the s,2 lines as you
like, i.e. s,3 might tell you the status of another server (I currently
have it giving me a report on 10 servers, just by using GoogleTalk).
:)
- --
Cheers,
Matt Riddell
___
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http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
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Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFE1Li/S6d5vy0jeVcRAkLxAKCGUhkZxwlTg56GswxOGXGQ3hdNPgCfYcf4
pdtc7b2C1fgE2uwf2k+imi8=
=A3g2
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Re: [asterisk-users] About Digium cards and HP DL servers
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: What do you want for the TE410P Give me a call! or e-mail me This is not the business list. In fact you will notice it states: Asterisk Users Mailing List - Non-Commercial Discussion Please stop spamming the lists. At what stage can someone do something about repeat offenders? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE019IS6d5vy0jeVcRAlOZAJ0d4inDrtdQlcspF7UyhRw2BwMUUgCfQU5k 5YuhufyGBElpUXIMb4C4/Ac= =kzuJ -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] VoipNow 1.2.0 Beta
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Does above means that the license for voipnow need to be paid to packet 8 as well? http://biz.yahoo.com/prnews/060613/sftu062.html Senad Hate replying on my post but what a heck!!! My understanding is that ANY hosted IP PBX coded in any object oriented programming language is falling under the above mentioned patent. Anyone has any thoughts on this? Another reason not to do business in the USA! :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEzlgdS6d5vy0jeVcRAmnkAJ9Ly1oYi+dClZ1EffkAFwXk6NOgoACeNzLr aKexFxd/ikEpONcncApH62Q= =b1Yb -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] SIP and podcasts
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dovid Bender wrote: Does anyone know if there is such a solution to listen to XM radio's service thru thier site ? What format is the feed? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEzLhVS6d5vy0jeVcRAr/ZAJ4kC7XFuQY2/CWGIju+KERc7ZMcWgCeO7sH PTtovIxfTWUbVHN48dOzQ44= =ASrU -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] VoipNow 1.2.0 Beta
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Man that looks nice. Kinda reminds me of the Plesk. Anyway, I've put up a screenshot with the original post at: http://www.sineapps.com/news.php?rssid=1399 - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEzChHS6d5vy0jeVcRAuB4AJ9M371l7B1JN/xFrp1OAdcqt/4h6ACeLKgT Jwxi5MvHoafqSumvzmNorTE= =hhpH -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] If you prefer to read this mail list as a forum ...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: On Thu, Jul 20, 2006 at 10:53:26PM -0400, augustynr wrote: Hi, I got realy tired of looking at Asterisk lists in Outlook so I moved it into the phpBB2 type forum. It seems to be working well for me and I think some of you may find it usefull too. So here it is at: http://forum.globalvoicenet.com/ One thing both MS-Outlook and phpBB have in common is the lack of decent threading support. This makes reading complex list threads much more complicated. Sadly, Outlook does not even preserve threading headers and thus its users force me to manually correct threading in the asterisk-users mailbox. Um, but aren't you using Mutt 1.5.9i? :) Oh maybe you mean they break it, you fix it! :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEzCidS6d5vy0jeVcRAn3xAJoDQd+HcEeX3RuY1oK+ZjfrSJUORgCfXHFe t4mFzrjzzs/GwP2agdvzFsg= =KqZu -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] SIP and podcasts
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 kael wrote: I find curious there's no solution to listen to podcasts via SIP servers. Um, I've just uploaded a total hack to the wiki: http://www.voip-info.org/wiki/view/PodCast You will need phpagi, the script, the extensions.conf entry, magpie rss and a bit of patience. Drop me a line in you have any problems! BTW: It is a total hack, I was planning to clean it up, but never got round to it, seeing as you're looking for it I thought I'd post it anyway. It works for me, but YMMV! :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFExq5vS6d5vy0jeVcRAvdOAJ9dbiEEp2MSj/SIZki29GSHCSAkaACfV0tZ hNm2DsK9j1vrAxJ+wg2RYg4= =IYze -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] question about asterisk DB
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 RR wrote: Unplug, I'm sure there are other people with better ideas but if you see on sineapps, I remember someone having written a patch which seperates out the the sip registry into a new table. If this is stable Save you searching: http://www.sineapps.com/news.php?rssid=1364 :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFExBN+S6d5vy0jeVcRAgupAJ9b7PVh5bqXX8P232vM/pUTpj2/xgCfQ5aD XqOTQb44gbxSHLxG6G0suFY= =vnEr -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] Sphinx and Asterisk Integration, How?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Zeeshan Zakaria wrote: I installed the packages listed on this webpage. But What is meant here by Server and Clients. Do I have to write these programs and then run then somehow, one on server and one on some client. From server I understand Asterisk box, but where is the client? The client is the agi script in Asterisk. Both the server and the client can reside on the same machine. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEvu0sS6d5vy0jeVcRAmvKAJ9GqGybtB8hrgUE2deak5NX94+1GwCfe7fR rgemAyj3+UXzJOgslhuQPA8= =XQg8 -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: Asterisk 1.2.10 and Zaptel 1.2.7 released!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Lacy Moore - Aspendora wrote: Can Digium legally sell this? It is my understanding that if the license of all parts cannot be legally sold, then there is no way it is going to be included. Well yes, if the licence is BSD like and all code is disclaimed to Digium. What's the bugtrack number for it? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEvu4XS6d5vy0jeVcRAvLFAJ9EfVRt1AvnyI4sKPG++W08vqe9/wCdGTdh ERqedg6Wwgg061apvsiqf00= =uGFk -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] Two security holes fixed in latest versions of Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: On Tue, Jul 18, 2006 at 10:13:58AM +1200, Matt Riddell (NZ) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 From: http://www.sineapps.com/news.php?rssid=1377 ISS Xforce has published details of two security issues in Asterisk 1.x which were fixed in the recently release 1.2.10 version. Asterisk IAX2 Protocol Denial of Service Attack Summary: ISS X-Force has discovered a denial of service vulnerability in the Inter-Asterisk eXchange protocol version 2 (IAX2). IAX2 is used by Asterisk PBX software to exchange Voice over IP call setup and call content. If an attacker floods the PBX with call requests, the PBX will be unable to handle new telephone calls. IAX2 Protocol Denial of Service Amplification Attack Summary: ISS X-Force has discovered a traffic amplification vulnerability in the Inter-Asterisk eXchange protocol version 2 (IAX2). IAX2 is used by Asterisk PBX software to exchange Voice over IP call setup and call content. An attacker can leverage accounts without passwords on an Asterisk PBX to flood a third party with a large amount of UDP packets. If the attack is properly constructed the amount of traffic generated can saturate the victim's Internet connection. Networks do not have to use Asterisk PBX to be the victim of this kind of traffic flood. If you wish to find more information and follow the links to ISS Xforce's site, you'll actually get irrelevant and misleading information. I remember the issue of amplification raised in the dev list a number of monthes ago regarding both SIP and IAX2. It is still not clear from those texts what version 1.2.10 has actually fixed here. Where can I find more details? Yeah, the amplification one is not fixed. The IAX2 DoS one is however fixed. As I'm sure you've noticed, there is some discussion on the methods to fix the second one going on in the dev list at the moment. Kevin's response is here: http://www.sineapps.com/news.php?rssid=1382 - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEvu6aS6d5vy0jeVcRAt5+AJ0WDpzVErzis5y/8BpaP4xu+/yjWgCfe3Pd 2cdEVdU8AeSob1S5+Ga+xjg= =gp4G -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] zaptel on dual processor, How?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Warren (mailing lists) wrote: Olivier Picquenot wrote: Zeeshan Zakaria a écrit : It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686 Then you might want to use yum to install the apropriate package, the one that contains the kernel source, or at the very least the kernel headers . Or you might grab it on a Cent OS mirror, for exemple: ftp://ftp.dedibox.fr/centos/4.3/updates/i386/RPMS/kernel-devel-2.6.9-34.0.1.EL.i686.rpm I'm no Cent OS expert, but that should be the right rpm . The proper method is, as root, type: yum install kernel-devel The problem is, the kernel headers will have the name 2.6.13-15.8 whereas uname -a will report 2.6.13-15.8-smp. You may need to create a symbolic link. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEvA0cS6d5vy0jeVcRApKfAJ9n9R+9jUt9Jh5A6oMZIIziClqSKQCfUxUS FOdt0/YMnnj0tIBnsbbZXBI= =8UHH -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
[asterisk-users] Two security holes fixed in latest versions of Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 From: http://www.sineapps.com/news.php?rssid=1377 ISS Xforce has published details of two security issues in Asterisk 1.x which were fixed in the recently release 1.2.10 version. Asterisk IAX2 Protocol Denial of Service Attack Summary: ISS X-Force has discovered a denial of service vulnerability in the Inter-Asterisk eXchange protocol version 2 (IAX2). IAX2 is used by Asterisk PBX software to exchange Voice over IP call setup and call content. If an attacker floods the PBX with call requests, the PBX will be unable to handle new telephone calls. IAX2 Protocol Denial of Service Amplification Attack Summary: ISS X-Force has discovered a traffic amplification vulnerability in the Inter-Asterisk eXchange protocol version 2 (IAX2). IAX2 is used by Asterisk PBX software to exchange Voice over IP call setup and call content. An attacker can leverage accounts without passwords on an Asterisk PBX to flood a third party with a large amount of UDP packets. If the attack is properly constructed the amount of traffic generated can saturate the victim's Internet connection. Networks do not have to use Asterisk PBX to be the victim of this kind of traffic flood. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEvAumS6d5vy0jeVcRAgO1AJ92+xi4BzBfGC7hQlAxVSOxJPFWPgCfcapd yfsmGcmGZE0LqinUJ5w16ls= =3lgI -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: Asterisk 1.2.10 and Zaptel 1.2.7 released!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 trixter aka Bret McDanel wrote: On Mon, 2006-07-17 at 19:21 +1200, Matt Riddell (NZ) wrote: It will sometimes tell you that there are modules inside /var/lib/asterisk/modules which were not compiled for the version you are compiling. If these are not asterisk-addons modules you will likely need to remove them. or modules from others that arent allowed to contribute to asterisk-addons or the tree itself for whatever reason, of which I have a few of those that have been specifically rejected for inclusion even though disclaimers are on file :/ politics at its finest. At least they work and it appears that some of them take less ram and cpu than default asterisk stuffs :) :) Which applications exist that have been disclaimed, well coded, are patent unencumbered and are not accepted? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEu9C8S6d5vy0jeVcRAuW9AJ0Z7+RC4+4sN6Sij0PySd9k2n0EmACeKAZx L5uAsnu61xqG0/tRXEDhuhE= =UMqE -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: Asterisk 1.2.10 and Zaptel 1.2.7 released!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Martin Joseph wrote: On Jul 16, 2006, at 11:12 PM, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... If it was a .tar.gz download then you will need to reinstall. Hi Matt! If I upgrade to 1.2.10 and than decide to go back to some prior version, how will I do that (using tar.gz)? I think if you keep the older source in a separate directory, you can always cd back to it and do a make clean, make, make install. This is only what I have gleaned from the list, so hopefully more knowledgeable list members will chime in. This is also the reason I have avoided building from SVN, as I like the idea of being able to revert to an earlier working build if need be... Also don't forget to pay close attention to the messages at the end of the make process when compiling and installing Asterisk. It will sometimes tell you that there are modules inside /var/lib/asterisk/modules which were not compiled for the version you are compiling. If these are not asterisk-addons modules you will likely need to remove them. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuzqBS6d5vy0jeVcRAqspAJ0enhDY0coXa2TjQOym25413CMotQCfb6+r e+s/AhF5yPREzBQmm6SnlOs= =sZVl -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] Asterisk 1.2.10 and Zaptel 1.2.7 released!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Julian Varanini wrote: So I can just install it over 1.2.9? This is what I did and everything seems to be working fine. Yes as long as it doesn't complain there are modules which were not compiled for the running version i.e. app_math - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuzs9S6d5vy0jeVcRAjsTAJ9vPYf14YAHwtZgO7JYxqqeYPYzoACfQm5f QdZ4fd8P1qhzjZyKEoUjMHA= =XBlw -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] SRTP enabling
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Martin Joseph wrote: On Jul 16, 2006, at 9:45 PM, Abdul wrote: Hello, In some countries i found that they are blocking SIP port 5060 so instead of this i change to another port 1221, and its work well. But in one country the are not blocking SIP but they are playing with RTP packets, if they filtered it is VoIP RTP they are doing something called party cannot hear or some time caller cannot hear but called party can hear well. So i cosider to use SRTP to make encryption. and i am using my asterisk in VPS so i have full control to manage the server. If you guys have better Idea to prevent such kind of issue, it will be good for us. Why not use IAX2? Then you only have one port to worry about reconfiguring Or alternatively run the whole thing over a OpenVPN UDP encrypted network (really simple to set up): http://openvpn.net/ - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuzuhS6d5vy0jeVcRAuYvAJ0UTWw2nZK+DWH8a9BE0w/klT8VpQCfSqd/ 07NexDPcXZsJA/t0VGFqZMA= =BcfJ -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] Injecting prerecorded audio into active call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nick wrote: Yeah a bit messy I guess. I had been hoping for a simple solution, but knew there most likely wasn't! The one idea I did have would be to use some kind of SIP api on the web backend. Then bring the backend extension into a conference, then from the web api you would have to select the call to play audio in. This idea would work well I think, as it would mean the system can be use regardless of the training call being active on the asterisk box, as long as their system supported conference calls. This is where I fall down though, I'm no developer! Anyone know of an api that would allow this? If you don't mind the call centre staff member pressing some buttons to request help in the middle of the call you could use a featuremap using features.conf and the playback application. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuzwjS6d5vy0jeVcRAuOLAJ9xBWUKiuFN2yLqxnnsYIXqig2XMQCfchOu 0EiFfyGOgOTwGSWxl2PrFwU= =luuK -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] Sphinx and Asterisk Integration, How?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Zeeshan Zakaria wrote: After several hours of searching the Internet, couldn't understand how can I integrate Asterisk with Sphinx voice recognition system. The sphinx software itself I've installed on my server. I need help from those who have successfully done it and can guide me how to do it. Thanks :-) Top link on Google: http://www.voip-info.org/wiki-Sphinx - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuzxpS6d5vy0jeVcRAmdwAKCJ+w19Tg+fbEacnymqhBCc+xDu4QCfY5SS Ksu18Wqeqt/eDVeWVpwDUSo= =Qhar -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] Manager action hold missing?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Wolfgang Falk wrote: Hello together, is it a fact that the manager api does not provide the hold and unhold commands? There is no manager command by that name nor is there any asterisk application by that name. You could maybe transfer a call to a parking extension using redirect and back again. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuLYbS6d5vy0jeVcRAvXFAKCLVTlSbZA8zcu8AQ1ZGf2e+moR0gCgjaSX m3FqzDnXprCgjtKnvKdFSUI= =WRK4 -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] Asterisk 1.2.10 and Zaptel 1.2.7 released!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Julian Varanini wrote: What is the best way to update from 1.2.9 to 1.2.10? If it was downloaded from SVN then you can just type make update in the directory. If it was a .tar.gz download then you will need to reinstall. I would recommend using the 1.2 branch of SVN as it means you don't have to wait for the releases to get the bugfixes. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuLm8S6d5vy0jeVcRAk9RAJ478UyMx8g7WLzkhAp+9VT9eZfXewCggHXo 9bn2Ob7u9jlDsqrKLZVrv/4= =y79J -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: Wrong account code from iax_buddies
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 voiplist wrote: I wish it were that simple.. We see the username coming in, it's in the channel etc.. We see the call come into one account and we see * set an account code for another account.. Really.. It seems that it has something to do with the fact that accounts registering from the same IP get mixed up. Anyone else experience this? We are using 1.2.4 on this particular box. Are you sure you don't have an account with no password or something? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuG+NS6d5vy0jeVcRAh/vAJ0cArD+Zs2fmKYmZZf+VumBVh0CUwCfRYa4 rywZhhYMlLFNWRCMc/4nrkw= =9gIG -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] Asterisk and VAD
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Abdul Lateef wrote: Hi all, does Asterisk 1.2.7.1 supporting VAD? because i am running my asterisk on VPS and i want to save badwidth. Not without timing patches, no. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuG/XS6d5vy0jeVcRAnXaAJ92bLRg+C7v+eCsyRJZU7b7RnAMqgCfXmzl lEu7IjfU9ILvDqiJ2I2ju4Y= =J2y7 -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] Inc.com Name s Mark Spencer of Digium to its “30 U nder 30: America’s Coolest Young Entrepre neurs”
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Randall H. wrote: Congrats Mark ! Link and excerpt here: http://www.sineapps.com/news.php?rssid=1366 - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEtonJS6d5vy0jeVcRAn7hAJ9hubkLGxQk/fnSiw6o6iw1y6mHLACfYWYy fri5OteCHuztxlTdlrm6sSY= =ZYPu -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] Server redundancy
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Asterisk realtime hardly provides redundancy. Can do. 1. There's no support for realtime SIP where multiple Asterisk systems can reference the same MySQL database for SIP peers. Ask Kevin Fleming about this. It's known not to work. Correct, although you can frontend with SER. 2. The IP address of the MySQL server is hard coded into the Asterisk config files. In the event of a database failure, Asterisk fails as well. You need to build redundancy into MySQL with a primary and seconday server, and something that can monitor MySQL system, network, and application and then transparently (to Asterisk, because it can't do it itself) switch IP's in the event of failure. If you use a MySQL cluster, the IP address will; be of the load balancer which will forward those requests to the MySQL database machines. Otherwise you can use http://linuxvirtualserver.org/ 3. Other stuff I can't recollect right now because I am tired. :) Aren't we all - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEtDM6S6d5vy0jeVcRAv+FAJ9n25jgD4tZz9rQeFlG5YER8Orh6ACZASy5 2phzn6IWuykDX2JyCC5mZOI= =rKxA -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] NuFone, please send the log file
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael Workman wrote: Very Simple. I hired JerJer to Have a SER and Asterisk setup with Acounting... JerJer told me to Talk to Shido6 and he would do it... He told me it Would cost me $3000 and he do it. He demanded the $ first and never did the work. How did you pay? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEtFHdS6d5vy0jeVcRAgocAJ4kbQR2CqMubcZDZBbju+w1q4GTxwCfUlax 3O3p8uz8SylNGGmj+vSh3+g= =gGrG -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] Numbered Voicemails when you still delete them.
Is there anyway, to delete a message, but still have some sort of incremental counter for the message id? Not that I am aware of. At least unless you script something yourself. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] french sounds in asterisk
serge messa wrote: Hi all i want to know where i can find french sounds for asterisk. I don't have any studio to register good sounds. Free sounds available now (recorded in a studio by my wife): http://www.sineapps.com/FrenchPrompts.tar.gz - ready to go in GSM format Also available: http://newton.waglo.com/~millette/asterisk/ - gsm, speex et ogg flac (Québec) http://www.westany.com - Professional French voice prompts that can be customised by the same voice artist. http://public.indigen.com/asterisk-1.2-sounds-fr-armelle.tar.gz - French female voice for asterisk 1.2 in GSM format -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] H.264
Erick Weber V. wrote: Hello: Does someone know if asterisk supports H.264 video codec Find attached cvs commit note, so yes, if you get the latest trunkversion of asterisk you will get h264. Subject: [svn-commits] trunk - r7855 in /trunk: ./ channels/ formats/ include/asterisk/ From: svn-commits@lists.digium.com Date: Sat, 07 Jan 2006 17:54:23 - To: [EMAIL PROTECTED], svn-commits@lists.digium.com To: [EMAIL PROTECTED], svn-commits@lists.digium.com Author: markster Date: Sat Jan 7 11:54:22 2006 New Revision: 7855 URL: http://svn.digium.com/view/asterisk?rev=7855view=rev Log: Add support for H.264 with SIP and recording -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Echo Cancelation on TE110P
Kerry Garrison wrote: On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? You can either run the software echo can or use a hardware one. You will have to enable it for all calls however. The echo will be coming from the remote system. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] can i get the script
pali ismail wrote: i have do some touch tones registration system in asterisk . know i hae some problem i my extensions.conf ,,,because the script there cannot run yet so i hope some budy have codeing in check password plase give to me i can check that my code its right or wrong :) You will need to show us the code. Is there any reason you are not simply using readdtmf or authenticate applications in the dialplan? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Oh this is bad.... bindaddr and rtp traffic
Douglas Garstang wrote: Pardon my candour, but for a product Digium calls 'enterprise grade' it sure seems to be missing a few features. Um...it's Open Source. Why don't you add the features you require yourself or pay someone to add them for you... This is your third similar post in as many days. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] MixMonitor Problems -- sssshh, don't be too loud
Brian Roy wrote: On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote: I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811.. I'm running 1.2.1 and most of mine get cut short too. I posted this on the list a few months ago and nobody had any suggestions. BJ said I should probably post a bug on it but I haven't had time to continue to troubleshoot it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been watching change logs and hadn't seen anything surrounding mixmonitor so I've let it go. Please continue to update us if anyone gets some resolution. I'm glad to know there are lots of us experiencing this. That should be the catalyst to get it fixed. The only catalyst to getting it fixed will be if someone posts a bug entry with full details on bugs.digium.com If you do, post again here with the ID and discussion and testing can continue there. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Oh this is bad.... bindaddr and rtp traffic
Douglas Garstang wrote: Asterisk calls the Business Edition 'enterprise grade'. It's right there on the Digium website. It's the same dang code as the open source version, just older. We are using it successfully in quite a few enterprise roll outs. If you are unable to, maybe you should attend one of our training sessions, which among other things discuss how to code for Asterisk. If however you'd rather just complain, please do so to /dev/null -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] res_mysql.conf DNS SRV lookup
Douglas Garstang wrote: Good grief! I posted the message below at 1:17pm... and it appeared on the list after 8pm. Nice I just posted mine and it arrived 30 seconds later...from New Zealand. Maybe your mail servers are b0rk3n: hehe :D It varies from time to time, but the mails do tend to go through! -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 text to display of sip phones
Alejandro Vargas wrote: I red that it is possible to send instant messages to the displays of sip phones. How can I do it using Asterisk? You can either do sendtext from an agi on that channel, or using my new patch ( http://bugs.digium.com/view.php?id=6131 ), you can do it from the manager interface just by passing the channel of a call. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] parking slot lights - testers wanted
Dr. Michael J. Chudobiak wrote: Hi all, The metermaid patch allows you to use the programmable buttons and LEDs on phones (like the Snom 2xx or 3xx) to view the status of parking slots and transfer to them. This should be really useful for small-office environments. Anyway, the patch seems to work with Snom phones (and hopefully others) now! The curious are encouraged to download the metermaid-v3.txt patch against v1.2.4 for testing and feedback! See http://bugs.digium.com/view.php?id=5779 for details. Is this the same one in the test-this branch? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 port mpg123 uses for MoH?
Zach A wrote: Hi, What port does mpg123 uses to play music on when it starts MoH after asterisk has put called on hold? As far as I'm aware it writes to standard output and reads from standard input (i.e. no ports involved) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???
Dan Miller wrote: So, when I get no comments on this at all, either here or on any of the forums, does that mean nobody knows what I'm talking about?? Or does nobody know the answer?? Or is it just a stupid question and nobody wants to bother telling me where to look?? It *is* a question that I have to answer somehow; I've read all through TFOT and see nothing relevant to this issue. It's silly to spend $15000 on a G723 license just so I can play back menu messages from Asterisk (since the actual call decoding is done by the external boxes, which have already paid the licensing fees). You can not really currently change codecs mid call (in most situations) although work has been progressing in this area for some time. Theoretically you should be able as others have based IAX devices around this concept, but I don't think its available for sip. Your other option would be to convert the audio files from GSM to G723.1 and that way, playing them would not require transcoding. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Professional Recordings
Waldo Rubinstein wrote: Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? http://www.digium.com/index.php?menu=product_categorycategory=thevoice -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 Video and Meetme
Hagen Rode wrote: Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video support to app_conference (also in iaxclient's CVS repository), and generally improve the iaxclient audio and codebase. So my guess from this is that there is currently no support for video in app_meetme, but that in the (hopefully not too distant) future, app_conference will be the replacement for app_meetme and will have video support. Replacement is a big word. I would expect that meetme in its current format will not be able to support video multiplexing. App_conference on the other hand looks like it may do. If you're into hacking code, the TIPIC libraries will support simple video communications at the moment, although I have been unable to compile it successfully. There was some discussion a while ago on the dev list regarding the replacement of meetme with app_conference, and the general consensus was that it wouldn't happen. This means that you will need to add it (possibly even in the future) to the apps directory and patch/alter the Makefile. OT: Mail me offlist if you are interested in building 3G/UMTS support into Asterisk. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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: G729 and Meetme
Martin Joseph wrote: On Mar 2, 2006, at 3:46 PM, Wai Wu wrote: You can really mix G729 encoded frames. So I would guess that licenses are not needed for non-G279 devices. BTW, there is a difference conference app (forgot the name) that only mixes the two parties that have the loudest volumn. It sounds more efficent to me this way. There is no reason to listen to three or more party talking at the same time anyway. I wish this was a joke. Sick and wrong is all I can say. :D Nah, iaxclient.sf.net has app_conference which does exactly that :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] test call quality
amaury BOSSE wrote: Is there a free linux tool which can test voip call quality between two Asterisk PBX. It will help me to test the WAN network between them. I have only found commercials ones, so if you know a free one, let me know. For packet loss, rtt etc and a phone call check out: http://www.sineapps.com/sinestatiax.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Meetme Timing Interface
Douglas Garstang wrote: I have ztdummy installed: Module Size Used by ztdummy 3464 0 zaptel218756 1 ztdummy crc_ccitt 2176 1 zaptel ohci_hcd 16388 0 floppy 49028 0 pcspkr 2180 0 piix8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 rtc10164 1 ztdummy usbcore84740 4 ohci_hcd,ehci_hcd,uhci_hcd However, when I enter a meetme conference, I get this: -- Playing 'conf-getconfno' (language 'en') Mar 3 15:27:26 WARNING[23657]: channel.c:2535 ast_request: No channel type registered for 'zap' Mar 3 15:27:26 WARNING[23657]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '123' Uhm WHY? If I didn't have ztdummy installed, Asterisk would complain that my conference number is not valid, and I would see errors on Asterisk startup about not being able to find a timing interface. These things are not happening. However, it is spitting out that error message on the console. Why? You need to compile asterisk after compiling zaptel. Otherwise chan_zap.so won't get created. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Lowering Server Load
Can you try not recording for a bit and see if that helps? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Changing caller id on transfer
Cosmin Prund wrote:
As usual, this is most likely a easy question, but here it goes any way:
How can I change the caller id on a transferred call so the called party
knows the call has been transferred from a colleague and it's not coming
directly from our outside lines?
The story goes like this:
1) Client calls. All phones ring.
2) Someone picks up the phone.
3) The phone gets transferred to someone.
4) The person that gets the transferred call sees the original caller id and
doesn't know the call has been transferred. I'd like the person that gets
the transfer to see the caller id with a digit prefix. Ex: Original
caller-id: 0269123456; Caller id if the call has been transferred:
1*0269123456
I know I can use SetCallerId(1*${CALLERIDNUM}) but how do I know I'm doing a
transfer and not calling someone?
You could do transfers for a number starting with 8 or whatever
So instead of transferring to 101 (the user's extension), you could
transfer to 8101. Then:
exten = _8XXX,1,SetCallerId(1*${CALLERIDNUM})
exten = _8XXX,2,Goto(extensions,${EXTEN:1},1)
Please not that the SetCallerID has been deprecated and should be
replaced in versions 1.2 with:
Set(CALLERID(number)=1*${CALLERIDNUM})
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Polling Asterisk for Life
Matt wrote: Yup.. that's the exact problem I'm having. I really can't explain what happens. If I don't restart asterisk it seems to happen after about 2 days. So I restart asterisk once a day at 3am. And it still goes down about once a month... Are you guys perchance using Local/[EMAIL PROTECTED] in your installations? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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
