[asterisk-dev] Centralized Conferencing Specifications in MeetMe
Hello all, we,Univeristy of Naples (Italy), are implementing a conferencing architecture based on IETF XCON working group specifics, at which we're collaborating too. Our choiceswere Asterisk as Multimedia Manager,a modified version of MEETME as conference mixer (audio at the moment) and some otherenhanced feature and, finally, a modified version of a SIP client called MINISIP. A first version isalready available and published as a sourceforge project at this address: http://sourceforge.net/projects/confiance basically we havestarteddeveloping some new feature for MEETME and modifying it in order to accomplish some new specifics. We add: 1.Moderation, implementing the Binary Floor Control Protocol (BFCP). 2. Conferences Control, implementing a custom protocol with a Scheduler component, waiting for the standard protocol. 3. Minor changes to accomplish WG specs. Any details are available at this address: http://confiance.sourceforge.net/index2.html We would like to continue our development according tothe communityso,any comments or feedback are welcome and, in particular, is it interesting for the community to have a branch with this code in ? Regards, Alfonso Buono - ~ Ing. Alfonso Buono CRIAI Consortium Tel.+390817766905 Mob.+393479609085 - ~ "La teoria è quando si sa tutto e niente funziona. La pratica è quando tutto funziona e nessuno sa il perchè. In questo caso abbiamo messo insieme la teoria e la pratica: non c'è niente che funziona...e nessuno sa il perchè!" Albert Einstein - ~ ___ --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
[asterisk-dev] Configuration for recording quality?
Hi. I have recently posted a mail on the users mailing list, asking around how to change the quality setting of files that asterisk record for you. For instance change the 8kHz for meetme recordings to 32kHz. The reply came that you can not configure the recording settings/quality. Is this true? I was just wondering if something like this is in the pipeline? Or was thought about? I was suprised to see that asterisk, which I regard as a functionality rich product, does not allow you to do this. Surely different poeple/companies/customers using asterisk in their voice/software solutions will have different quality requirements. Thanks. Regards, Jan. ___ --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] Configuration for recording quality?
On 8/4/06, Jan du Toit [EMAIL PROTECTED] wrote: Hi. I have recently posted a mail on the users mailing list, asking around how to change the quality setting of files that asterisk record for you. For instance change the 8kHz for meetme recordings to 32kHz. I don't think there would be any point (that is to say, any quality improvement) in recording at anything other than 8KHz at present, since all audio passing through Asterisk is sampled at 8KHz anyway. If you need 32KHz files for compatibility with another application, it should be fairly trivial to upsample the recorded files using a utility such as sox. -Rusty ___ --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] Configuration for recording quality?
snip On 8/4/06, Jan du Toit [EMAIL PROTECTED] wrote: Hi. I have recently posted a mail on the users mailing list, asking around how to change the quality setting of files that asterisk record for you. For instance change the 8kHz for meetme recordings to 32kHz. I don't think there would be any point (that is to say, any quality improvement) in recording at anything other than 8KHz at present, since all audio passing through Asterisk is sampled at 8KHz anyway. If you need 32KHz files for compatibility with another application, it should be fairly trivial to upsample the recorded files using a utility such as sox. -Rusty There was a discussion about having the audio passed through * be at a higher rate than 8k, thinking behind it was that the 8K limit is imposed by the PSTN and VoIP does not exibit that inherent limitation. I do not know of any endpoints that support a higher rate (8k), it would be nice to be able to use a High-Quality MeetMe room for a podcast-type live discussion. Will the incorporation of Video into a MeetMe room change this at all?? ___ --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] Configuration for recording quality?
On Fri, 2006-08-04 at 13:42 +0200, Jan du Toit wrote: I have recently posted a mail on the users mailing list, asking around how to change the quality setting of files that asterisk record for you. For instance change the 8kHz for meetme recordings to 32kHz. Unfortunately, this list isn't the I asked on the -users list, but nobody answered list. This is not a court of appeals. This is the list the Asterisk developers use to discuss changes to the Asterisk source code. The reply came that you can not configure the recording settings/quality. Is this true? As far as I know (and I could be wrong here), it's not possible. Why, you ask? First of all, because Asterisk deals with 8kHz audio, because that's what comes across the PSTN. While there is certainly VoIP hardware (and softphones) that support wideband audio, it's not that mainstream yet. [1] I was just wondering if something like this is in the pipeline? Or was thought about? Yes, the developers have thought about it, and I think you'll see Asterisk start to support wideband audio more and more once Asterisk 1.4 has been released and has a chance to settle. In fact, I think at least one of the developers has a wideband tree in the subversion repository, although I haven't seen any movement on it in a while, most likely because the developers are busy trying to get the finishing touches on 1.4 done. [2] I was suprised to see that asterisk, which I regard as a functionality rich product, does not allow you to do this. At this point, what good would it do to be able to provide a 32kHz recording of a meetme conference, if the audio coming into it is only 8kHz to begin with? If you really want 32kHz audio, why not use sox to convert the file after the fact. I apologize this this comes off sounding rude -- I don't mean it to be that way. I'm just trying to explain things in a way that will help you understand. -Jared [1] See chapter 7 of Asterisk: The Future of Telephony for an explanation of why it's 8kHz. The PDF is downloadable for free from www.asteriskdocs.org [2] http://svn.digium.com/view/asterisk/team/mattf/asterisk-wideband/ ___ --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] res_jabber.c and an empty parser
We've had a few core dumps over the past couple of days. After looking with gdb, they all seem related to the following: #14 0x0041b27d in iks_send (prs=0x0, x=0x9e74974) at stream.c:499 #15 0x0040d978 in ast_aji_send (client=0x9c50b50, address=0x222b8a9 [EMAIL PROTECTED]/asterisk, message=0x222b8d2 this is a message) at res_jabber.c:1277 Notice that the prs pointer in iks_send is 0x0 - this is passed to iks_send by ast_aji_send, using the client-p pointer. This indicates that client-p (the jabber parser) is invalid. How can this happen ? We are not reloading, or restarting anything (servers / IM servers / etc). I have at least 5 cores with the same information. (a) #14 0x0096227d in iks_send (prs=0x0, x=0x9c7a35c) at stream.c:499 #15 0x00acd978 in ast_aji_send (client=0x9a80b00, address=0x1ed9d79 [EMAIL PROTECTED]/asterisk, message=0x1ed9da4 this is a message) at res_jabber.c:1277 (b) #15 0x0041b27d in iks_send (prs=0x0, x=0x8979574) at stream.c:499 #16 0x00180978 in ast_aji_send (client=0x869d410, address=0x2328d79 [EMAIL PROTECTED]/asterisk, message=0x2328da0 this is a message) at res_jabber.c:1277 etc etc We could, of course, simply add a check to see if the client-p is valid before trying to send the message. That would stop the core dump, but I would like to know why it is happening in the first place. Should I raise a bug ? Julian. ___ --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
