Re: [asterisk-dev] move zaptel kernel drivers to a subdirectory
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Someone needs to change the HPEC installation instructions. Took me a while last night to figure out what was going on :) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHqfgmDQNt8rg0Kp4RAosBAKC9QgBOCLq1NXMOkK30VpKnBoU6PwCeO4fu CYC7sQsm+6MkN7sV5ObZ/QA= =8oHL -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] AGI and DeadAGI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 This is just a quick query regarding the latest change to report an error if DeadAGI is used on a live channel. (New Revision: 75437) If you have an AGI that you need to keep running after the channel has terminated (I.E. routing and billing), I was under the assumption that DeadAGI was the better to use. - From VoIP-Info: == The main difference is that the AGI application *may terminate* if the line is hung up during execution and DeadAGI will not terminate even if the call is hung up during execution, however, the call leg will not automatically enter a down state until execution is completed if executed on a live line. As such, commands and applications designed to return the call state will inaccurately return an up status. == The latest version states: Running DeadAGI on a live channel will cause problems, please use AGI So, what are we supposed to use if we want to run an AGI on a live channel and continue processing after hangup? I know this is -userish but I was hoping juggie might be able to respond as he wrote the patch. Are we fine using AGI in this situation? If so, I'll update the wiki to be a bit more verbose. - -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGpXV8DQNt8rg0Kp4RAlYhAJwMfUBwtDpra7mcsTo/E/vcPKAv1ACfZNGt WBvlqVsgV5ArvC04I8dSqfU= =y8oZ -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Problem building Zaptel 1.4 from SVN
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 The first patch worked, but the second didn't. By that I mean that applying the changes from the first one I was able to compile, but the second one where we just check for spinlock did not work. Maybe the problem is that this is a Fedora Core 3 system. Regardless without the #if...#endif I was able to compile without problems. - -- 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 iD8DBQFFzUEzS6d5vy0jeVcRAkq9AKCEMjMCPMBYNFEfBA0uJHEeb3lkWwCdGFA8 rCXxuNWwGSmJOFSQFs+VI38= =ZaoN -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk, Asterisk-Addons, Zaptel and Libpri 1.4 betas released!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rod Dorman wrote: On Thursday, September 21, 2006, 15:14:02, Asterisk Development Team wrote: The Asterisk development team is very pleased to announce that we have released the first 1.4 beta packages of all four of our projects! The beta versions are: Asterisk - 1.4.0-beta2 (beta1 was not released to the public) Asterisk-Addons - 1.4.0-beta1 Zaptel - 1.4.0-beta1 libpri - 1.4.0-beta1 Whatever happened to 1.3? I like the -beta# nomenclature, its similar to the Postfix a.b-mmdd naming of beta releases but aren't people going to ask where 1.3 went to? 1.3 was trunk once 1.2 was released. 1.5 is now trunk. As far as I understand. - -- 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 iD8DBQFFE5d2S6d5vy0jeVcRAql2AKCJR4RCDgaIQ/03KGLRJ3SUfCll+QCfX7x+ cr6SxzARaidRNGSqBiYbF+M= =Anrt -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] func_math.c
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kai Ober wrote: at least i have done it :)) plz look here http://bugs.digium.com/view.php?id=7959 i included math.h used pow and after that i mentioned that i could use LSHIFT as well so i implemented L and R shift as well PLZ have a look, and lemme know if you see some issues with too big values for num1 or num2 better ideas for op? pow = P and lshift =L??? Looks good to me. I'd expect this would be considered a new feature, but is such a simple patch, I'm not sure. Code looks clean, no style problem. - -- 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 iD8DBQFFCptIS6d5vy0jeVcRAm8RAJ9elIh1ZA7xnMVaBY1snci53cf7GgCfQgl6 DOebzwmVkRmcoXxm1O71rc8= =bHyN -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk Appliance?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Arnd Vehling wrote: Is there any info about the mentioned Asterisk GUI framework available? Not as of yet. Digium has said that while there are multiple Asterisk GUIs around, they feel (rightly so) that they understand the Asterisk codebase very well, and that they will therefore be creating (or are in the process of creating) their own GUI. I will post information as I find it. - -- 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 iD8DBQFFCFxiS6d5vy0jeVcRAr8NAKCMEZSx0NtMjaGtYDVAteCjFXyCqQCghzLy UFDljSv8UaW0gLjbNSuECW8= =lwE0 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Copyright issues with libcurl and OpenSSL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 James Jones cross posted. DO NOT CROSS POST. THIS HAS ALREADY BEEN ANSWERED IN ASTERISK-USERS!!! - -- 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 iD8DBQFFCIfyS6d5vy0jeVcRAiLXAJ9VUn1uX8syuvtz9FwyzM8I3A8a/gCeL0GZ rPChoDdtUNvA0SrDP/o7z8g= =6Miw -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] digium closing the source?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Roy Sigurd Karlsbakk wrote: hi all it's been some time this question came up, but i just heard a rumor saying 'digium is to be closing the asterisk++ source'. the rumors didn't say when or anything, so i'm just wondering if there might be something in this, or if it's only talk. I too would like to know where you heard this or whether you just made it up... - -- 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 iD8DBQFE+zRkS6d5vy0jeVcRAr9dAJ9fdMnYYJi1vdjNgFX3hUSRI+/wMgCfVTiK PaSgVr1bAMmuSe7L5kvBc1A= =ELc3 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] crash when entering WaitExten with moh class specified
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Pavel Jezek wrote: Asterisk SVN-trunk-r41849 kernel 2.6.17-2mdv when entering application waitexten _and_ if moh class name is specified, asterisk crashes, waitexten without moh class, e.g. simple WaitExten(10|m), is working fine (but no moh is played) Author: bweschke Date: Sun Sep 3 15:44:14 2006 New Revision: 41916 URL: http://svn.digium.com/view/asterisk?rev=41916view=rev Log: Fix enum indexing problem with m() in WaitExten. Reported by Pavel J, in asterisk-dev. - -- 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 iD8DBQFE+0Q0S6d5vy0jeVcRAmLLAJ9kS51ZiLs5lIyn/g1BtpTqCrtgYwCdF5cE MLcj54onC70zDJyYsQQ0oT4= =PnY+ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] OT: RSS Feed for bugs page (Mantis)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 This is a little off topic, but does anyone know if there is a link to receive an RSS feed of the changes to the bugtracker? I know there is one for the comments front page, but it would be really cool to be able to see when there are new bugs or changes to existing ones, even if just to see where the activity is and to pass on requests for testing to the community. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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 iD8DBQFE+2BSS6d5vy0jeVcRAk45AJ0RXCKnnmQ6i1Vm6qzUt3B1h4snGACfZI41 vUw3hKDWUSMwwVhmQu8OaIc= =Psvu -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] OT: RSS Feed for bugs page (Mantis)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Matt Riddell (IT) wrote: This is a little off topic, but does anyone know if there is a link to receive an RSS feed of the changes to the bugtracker? I know there is one for the comments front page, but it would be really cool to be able to see when there are new bugs or changes to existing ones, even if just to see where the activity is and to pass on requests for testing to the community. Sorry about the noise: http://bugs.digium.com/issues_rss.php :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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 iD8DBQFE+2KcS6d5vy0jeVcRAmlvAJoCC5ScZPTXgibvpBf4TBGf6N44PgCdGW/T U1CA8I7HF8H+f+6iRrQJ5Ek= =CNkp -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Jabber Manager Command
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I'm trying to let the manager send jabber messages and I include jabber.h then I have: struct aji_client *client; and later client = ast_aji_get_client(jabber); but I get the error: usr/src/asterisk/manager.c:889: undefined reference to `ast_aji_get_client' it's listed in jabber.h and implemented in res_jabber.c it doesn't die till it does the ld part any ideas? in the .h file it has: struct aji_client *ast_aji_get_client(char *name); and it gets implemented fine in res_jabber.c I'd really like this to work, and I'm pretty sure it's a minor screw up on my part :) Any help would be much appreciated! :) - -- 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 iD8DBQFE1ggUS6d5vy0jeVcRAqiwAJ91+ZLH5fD3VvwVpHqx6tTJci6fJwCfak8B 4SwL/6pLTP0tVVttoXWuTx8= =gZmQ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Jabber Manager Command
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Russell Bryant wrote: On Sun, 2006-08-06 at 17:17 +0200, Matt Riddell (NZ) wrote: I'm trying to let the manager send jabber messages usr/src/asterisk/manager.c:889: undefined reference to `ast_aji_get_client' The problem is that this function is not implemented in the Asterisk core. It is implemented in a module. It is saying You used this function, but I can't find it.. any ideas? There are some tricks that would make this possible to implement in manager.c, but there is a better way. Implement this manager action is res_jabber.c. See other modules, such as something like chan_sip.c., to see how it registers and unregisters manager actions when the modules is loaded and unloaded. Functions of interest when looking up how to do this are: ast_manager_register2() in load_module() ast manager_unregister() in unload_module() Cool, thanks for 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) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE1hbOS6d5vy0jeVcRAjJOAJ9q8YwmE+nba/leNVMEW/MqFCCKiACfcQDA E0U/VwJRRP8jkk8DI2pj7bo= =znwn -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] 'IAX2 call variable passing between servers '
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dave Cotton wrote: You have 2 choices: 1) Do the work yourself 2) Pay for someone to do it for you - -- Cheers, Matt Riddell No Matt you've got it wrong Decartes didn't say Je pense donc je suis he said Je râle donc je suis. :) - -- 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 iD8DBQFE0zFlS6d5vy0jeVcRAmA2AJ4+NnMeEDJRUuMgWRMph0wI5yOs9ACfUnoP ka91fvMpkU2JNNmlzVjpQdM= =DUTl -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] MeetMe without Zaptel or ztdummy
Anthony LaMantia wrote: Hi, Has anyone ever tired to design a conferencing application or tweeking a existing one, such as app_meetme to not require to use zap for trunking/bridge operations? using /dev/zap/pseudo seems to be fine for most causes with ztdummy and all, im just wondering.. Have a look for app_conference. Hint: iaxclient.sf.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) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] losing CDRs on mysql backend
Juan Pablo Abuyeres wrote: Hi, I am losing CDR records on my MySQL backend. I just compared CSV records with MySQL's and I was surprised with the difference. Searching a bit, I hit this bug: http://bugs.digium.com/view.php?id=4953 Questions: 1.- kpfleming says The spooling issue will continue to be a problem for v1-0 users, but HEAD (and v1-2) users can use the built-in spooling in the CDR engine.. That built-in spooling has to be enabled somewhere or is it automatic? 2.- If there is a built-in spooling into CDR itself and there is no spooling in cdr_addon_mysql, when the mysql server is down, crashed, full, unwired, firewalled, or whatever.. is built-in spooling in CDR itself supposed to handle that?? I'm using asterisk-1.2.7.1 and asterisk-addons-1.2.2 Is there any way that we can tell which cdr records have been lost? Is there an error in the console? Does this apply to all realtime cdr operations i.e. postgres, odbc etc or just MySQL? -- 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-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Asterisk Port to Windows
Chad Brown wrote: Does anyone know if an effort is being made to port Asterisk to Windows? I’m not talking about Cygwin but a native port. http://www.asteriskwin32.com/ But I'm not sure about the trademark usage in the domain name unless they sorted something with Digium. -- 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-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] res_config_mysql.c connection problems = bug
Loic DIDELOT wrote: Hello, I have for every registration about 3-5 selects and the same thing for every call. But I only have one update per registration so every 5 minutes I have been able to do 5000 update in 3 seconds on my mysql server. Cool, have you got a bug tracker id for the patches? -- 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-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Require Help in Detection of Human Voice or Answering machine on Called telephone number
Muhammad Asim Sajjad wrote: Hi, i am very thankfull to you on this helping material but i require more help in this regard. My problem is that i want to know how can i detect that the Human of Answering machine is connected with my called phone number. BackgroundDetect or MachineDetect - really a *-users question though. -- 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) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe
Tilghman Lesher wrote: On Thursday 10 November 2005 03:38, Chih-Wei Huang wrote: Chih-Wei Huang wrote: BTW, my G.723.1 and G.729 codec are from Intel IPP library. I decided to file a report to Mantis: http://bugs.digium.com/view.php?id=5697 Hopefully it can be fixed before 1.2.0 release. Due to patent issues and the legal problems associated with contributory patent infringement, we cannot do anything about any problems you're having with the G.723.1 codec (at least until the associated patents expire). Which is when? Do you really see them allowing the patents to expire? -- 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) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] chan_exosip2 for asterisk
Jerris, Michael MI wrote: Matt Riddell harry gaillac wrote: Hello, Is it possible to provide chan_exosip2 for both asterisk and openpbx ? Only if the authors are happy to disclaim the code to Digium. Disclaiming chan_exosip would make little difference as it is based on GPL libraries and therefore can not be part of asterisk itself. It could be included in addons, and to my understanding would not require a disclaimer for that. Whoever wrote the GPL libraries can probably still disclaim them. But yeah as I understand it, it could be included in addons. -- 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) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe
[EMAIL PROTECTED] wrote: On Mon, 14 Nov 2005, Matt Riddell wrote: Tilghman Lesher wrote: On Thursday 10 November 2005 03:38, Chih-Wei Huang wrote: Chih-Wei Huang wrote: BTW, my G.723.1 and G.729 codec are from Intel IPP library. I decided to file a report to Mantis: http://bugs.digium.com/view.php?id=5697 Hopefully it can be fixed before 1.2.0 release. Due to patent issues and the legal problems associated with contributory patent infringement, we cannot do anything about any problems you're having with the G.723.1 codec (at least until the associated patents expire). Which is when? Do you really see them allowing the patents to expire? Patents expire whether you allow them or not. It is kinda tricky for g.723.1 - there are lots of patents, filed in different jurisdictions. Some of them in fact have expired already (2004). I saw a statement that all of them will expire by end of 2006... Cool, so it's not like the statute of limitations where you can make changes and start the clock again? Any word on g729? -- 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) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] [OTAnn] Feedback
shenanigans wrote: I was interested in getting feedback from current mail group users. This is really pushing it now. That's three times. -- 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) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Asterisk or Polycom Bug?
Martin Mateev wrote: it's friday night, at least here, get a life Um...it's Sunday here (5:13am) and I'm still working. -- 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) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback
Kevin P. Fleming wrote: Matt Riddell wrote: Sorry, where is this Kevin? Look for VM_CATEGORY in the source, it's a channel variable. I don't see anything related to it in voicemail.conf.sample as of 30 seconds ago :) It's not in there, because it doesn't belong there :-) If it was a configuration option, it wouldn't be per-call selectable G :) I was expecting something like usecategories=yes :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Help IP phone project
[EMAIL PROTECTED] wrote: Very true. Too bad that there is no manufacturer interested in producing a non-ghetto looking PA168 phone. All I need is some blinkenlighten and softkeys, and rest can be managed with software... How many units you looking at? It could be cost effective to make something custom if enough people were interested. Any votes for an Open Source development of a phone with Open Source firmware? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Sending a REINVITE to a SIP channel
Juan Jose Comellas wrote: For some applications I'm building on top of Asterisk sometimes I need to redirect a SIP channel to another server. I intended to do this via a REINVITE, but I haven't found an easy way to do this from an Asterisk application. Is there any way to do this? What about dial? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] [Rant] [long] - code style and quality
Michael Giagnocavo wrote: Yea, I'd recommend people to read Code Complete (now in second addition) that would back up a lot of what you say with hard data from a lot of real projects. I'd second that recommendation even though the book came out of the Microsoft Press! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] asterisk stable?
Roy Sigurd Karlsbakk wrote: frustrated Hi running 1.0.6/1.0.7, I keep seeing numerous bugs such as huge memory leaks, rtp overflows, application bugs etc etc. Wasn't the 1.0.x track meant to be stable? Or does stable only mean feature freeze? Will asterisk 1.2.x be just as badly tested as the 1.0.x track? roy /frustrated What kind of response do you expect to that? Why not just post the bug id's of the bugs you have found (preferably with patches). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Re: FW: asterisk/apps app_meetme.c,1.71,1.72
Brian West wrote: I just spoke with mark and we are all gonna get together and come up with MeetMe2 (it will be in addons for a while) *PHEW!!* (The sound of a collective sigh of relief) :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] astcc
Darren Wiebe wrote: A patch for the divide by zero bug is sitting in bugs.digium.com waiting for some sound files. Somebody want to volunteer to finance a few files? Or does somebody have a decent sound editing setup and knowledge? You mean editing or recording? I don't really have the time to go down to the studio at the moment, but I've got a full on editing suite at home. I've been doing the sound engineering/recording/mastering thing for like 15 years now so I guess I have some knowledge. At least I'd hope so! :-) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] New card - TE110P?
Anton Tinchev wrote: Matt Riddell wrote: Anton Tinchev wrote: Will be there new card? I'm asking it, 'couse i'm going to buy 3-4 cards? Or i should wait for the new one? I'd buy now. There has so far been no information on the new cards from Digium (I.E. no confirmation that the card will exist and consequently no release date) and the current cards have been proven to work well. The problem i that i'll going to experiment with both E1 and T1 equipment. A single span E1/T1 changing card will cost half for me :) In that case I'd recommend the quad span E1/T1 card which is available now. (also seeing as you stated you needed 3-4, this will be a lot cheaper) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Delay/latency on SIP phone with Asterisk
[EMAIL PROTECTED] wrote: Hi folks, Does anyone have any suggestions for how to reduce latency in Asterisk? I have just a small hobby system running on a PII 400 MHz with a SIP (X-Lite) softclient and a SIP hard-phone (Integrated Networks IN1002 - PA1688-based phone). I am experiencing a significant delay - perhaps 500 ms or so - on the Echo test and when one client calls the other, which is obviously not good. I am running the Asterisk 1.0 release on Fedora Core 2 (2.6 kernel). I've been through the various articles on the WIKI without much joy. Thanks! Wrong list. This is an asterisk-users question as you are wanting to _use_ Asterisk. 1) Turn on qualify (i.e. put qualify=2000 in sip.conf for each entry) and then you'll be able to see which is causing the delay. 2) Try out an IAX softphone... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] New card - TE110P?
Anton Tinchev wrote: Will be there new card? I'm asking it, 'couse i'm going to buy 3-4 cards? Or i should wait for the new one? I'd buy now. There has so far been no information on the new cards from Digium (I.E. no confirmation that the card will exist and consequently no release date) and the current cards have been proven to work well. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Busycount/Busydetect
You'll have to forgive me if this seems like a stupid question, but I'm more than a little drunk. Does the Busycount=8/Busydetect=yes etc actually check for the length of tone defined in indications.conf? Reason being that it seems it is hanging up lines if it receives 8 x the_tone_specified as opposed to 8 x the_tone_specified x the_length specified. Kind regards, Matt Riddell CEO http://www.sineapps.com ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Interesting fix on CVS
Update of /usr/cvsroot/asterisk/channels In directory localhost.localdomain:/tmp/cvs-serv27553/channels Modified Files: chan_vpb.c Log Message: / check so as not to enable loo-drop on FXS ??? A loo drop on FXS? Doesn't sound too nice! Matt Riddell ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] res_data Issue
On 10 Aug 2004 at 17:09, Robert Jackson wrote: Asterisk Development Gurus, As a bit of background, I just finished writing a patch to res_data to make the MWI work with chan_sip. It seems to work pretty well with chan_sip, and I am waiting to see if I should do the same for chan_iax2. Of course I immediately thought that my changes were causing this issue, but after I rolled them back the problems were still occurring. I am having a bit of problems using res_data with CVS as of 2004/08/09. res_data is working perfectly in our production environment with CVS 2004/07/14. Basically, once a user has been authenticated by app_voicemail no matter how they hangup * says Killed and completely exits. I did a bt on the dump file and here is the output: (gdb) bt #0 0x40023f92 in pthread_mutex_lock () from /lib/libpthread.so.0 1 #0x40179e8d in free () from /lib/libc.so.6 2 0x44342514 in #vm_execmain (chan=0x8173ed8, data=0x450286d4) at app_voicemail.c:376 #3 0x080733ef in pbx_exec (c=0x8173ed8, app=0x8133808, #data=0x4502b844, newstack=1) at pbx.c:469 #4 0x0807b599 in pbx_extension_helper (c=0x8173ed8, context=0x8174030 from-sip, exten=0x8173ed8 SIP/2302-d80c, priority=1, callerid=0x1 Address 0x1 out of bounds, action=135087704) at pbx.c:1383 #5 0x08075428 in ast_pbx_run (c=0x8173ed8) at pbx.c:1878 #6 0x0807bdf1 in pbx_thread (data=0x40023f80) at pbx.c:2097 #7 0x40023041 in pthread_detach () from /lib/libpthread.so.0 #8 0x401cbb7a in clone () from /lib/libc.so.6 (gdb) Since I am only getting into asterisk programming I haven't been able to figure out how to fix it. If I remove optimization from the Makefile * no longer crashes and I receive the following on the console: free(): invalid pointer 0x4202f654! After looking into app_voicemail.c it seems that the free_user method is whats throwing this error. My assumption is that the parameter (*vmu) is NULL or something, but if it were null it wouldn't be executing the free() method call. I did try to run the latest CVS without res_data on our test box, and the issue did not crop up which leads me to believe that it is an res_data issue not an asterisk issue. At this point I have tried my damndest to figure this out, but at this point I am just running into a wall. Are you using Linux 2.6? If so try changing the kernel stack block size from 4K to 16K, this has helped some people. Matt Riddell ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
