Re: [asterisk-users] Codec negotiation issue (no audio format found to offer)
- Original Message - > From: "Ryan McGuire" > To: asterisk-users@lists.digium.com > Sent: Wednesday, August 3, 2011 9:47:42 AM > Subject: Re: [asterisk-users] Codec negotiation issue (no audio format found > to offer) > From looking into this, it appears as if this is due to Asterisk > negotiating the legs separately as if they were not related to the > same call. So the ingress leg negotiates ulaw, and despite it knowing > that the peer also supports g729 fails the call since it's already > decided on ulaw and the egress leg only accepts g729. > > > If this is design intent I'm wondering if there is demand enough to > justify a feature request? > > > Any advice on how I can work around this issue? > > > Thanks, > > > -Ryan This is a much more complicated issue than Asterisk negotiating each call leg separate of one another. Even if we give one call leg information about call setup occurring on the other call leg it is about to be bridged to, we are dependent on the endpoints honoring the codec preference priority we send them to avoid translation between one codec and another during the bridge... Honoring the preference order in the SDP does not always occur, which means that any effort in this area would only work in a perfect scenario. Even if we get call legs to negotiate perfectly before being bridged during call setup, we are not guaranteed that translation will not occur later if the call is transfered or parked. Regardless of what we do, if your setup allows ulaw and g729 for sip peers, you will always run the risk of a codec mixmatch... You may however be able to avoid this for some cases by using the sip.conf preferred_codec_only option. I believe that will limit the codecs negotiated during call setup to the single codec currently chosen on the other call leg. The problem with this is that we are not guaranteed the call leg supplying the codec will not change later. -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_gtalk load error
- Original Message - > From: "--[ UxBoD ]--" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, July 18, 2011 11:42:25 AM > Subject: [asterisk-users] chan_gtalk load error > Hi, > > When starting Asterisk (1.8.5.0) I see in messages: > > [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module > 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined > symbol: ast_aji_get_client > [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' > could not be loaded. > > Yet I do have iksemel installed: > > ls -l /usr/local/lib/libik* > -rw-r--r-- 1 root root 281994 Jul 18 16:14 /usr/local/lib/libiksemel.a > -rwxr-xr-x 1 root root 822 Jul 18 16:14 /usr/local/lib/libiksemel.la > lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so -> > libiksemel.so.3.1.1 > lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so.3 > -> libiksemel.so.3.1.1 > -rwxr-xr-x 1 root root 165132 Jul 18 16:14 > /usr/local/lib/libiksemel.so.3.1.1 > > and checking whether they have been linked okay: > > ldd chan_gtalk.so > linux-vdso.so.1 => (0x7fff01523000) > libiksemel.so.3 => /usr/local/lib/libiksemel.so.3 (0x2b6fbeb09000) > libssl.so.6 => /lib64/libssl.so.6 (0x2b6fbed15000) > libcrypto.so.6 => /lib64/libcrypto.so.6 (0x2b6fbef62000) > libpthread.so.0 => /lib64/libpthread.so.0 (0x2b6fbf2b3000) > libc.so.6 => /lib64/libc.so.6 (0x2b6fbf4ce000) > libgnutls.so.13 => /usr/lib64/libgnutls.so.13 (0x2b6fbf827000) > libgssapi_krb5.so.2 => /usr/lib64/libgssapi_krb5.so.2 > (0x2b6fbfaab000) > libkrb5.so.3 => /usr/lib64/libkrb5.so.3 (0x2b6fbfcd9000) > libcom_err.so.2 => /lib64/libcom_err.so.2 (0x2b6fbff6f000) > libk5crypto.so.3 => /usr/lib64/libk5crypto.so.3 (0x2b6fc0171000) > libdl.so.2 => /lib64/libdl.so.2 (0x2b6fc0396000) > libz.so.1 => /usr/lib64/libz.so.1 (0x2b6fc059b000) > /lib64/ld-linux-x86-64.so.2 (0x003ac420) > libgcrypt.so.11 => /usr/lib64/libgcrypt.so.11 (0x2b6fc07af000) > libgpg-error.so.0 => /usr/lib64/libgpg-error.so.0 (0x2b6fc0a21000) > libkrb5support.so.0 => /usr/lib64/libkrb5support.so.0 > (0x2b6fc0c25000) > libkeyutils.so.1 => /lib64/libkeyutils.so.1 (0x2b6fc0e2d000) > libresolv.so.2 => /lib64/libresolv.so.2 (0x2b6fc102f000) > libselinux.so.1 => /lib64/libselinux.so.1 (0x2b6fc1245000) > libsepol.so.1 => /lib64/libsepol.so.1 (0x2b6fc145d000) > > Any thoughts on why this is happening as I could not find many > references to it when searching ? > -- > Thanks, Phil > Do you have res_jabber installed? -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
- Original Message - > From: "Chris Maciejewski" > To: asterisk-users@lists.digium.com > Sent: Friday, May 20, 2011 8:56:35 AM > Subject: Re: [asterisk-users] ConfBridge - Failed to find a bridge technology > to satisfy capabilities > > Attach a debug[1] log so we can see what is happening. > > > > [1] > > https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information > > debug logs below: > > Asterisk 1.8.4: http://pastebin.com/DFnKgSse > Asterisk trunk r319661: http://pastebin.com/B19tdbxJ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users Just as a note, the versions of ConfBridge in 1.8 and Trunk are completely different. Trunk will likely give you much better results. In 1.8 ConfBridge is more of just an experimental exercise of the bridging API. -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new confbridge
- Original Message - > From: "Jerry Geis" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, April 25, 2011 10:17:41 AM > Subject: [asterisk-users] new confbridge > Is the new conf bridge going to be in 1.8? or only 1.10? > > Jerry > It will be introduced in Asterisk 1.10. Please don't create a new thread for questions regarding the new ConfBridge application when there is one already going on. ~David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!
- Original Message - > From: "David Backeberg" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, April 25, 2011 9:49:05 AM > Subject: Re: [asterisk-users] The new ConfBridge application is now in > Asterisk Trunk! > On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich > wrote: > > > > > > Does this ConfBridge requires a hardware timing source? > > No, and neither does MeetMe with modern DAHDI. > > > Will I be able to use this on any virtual server without having the > > need special changes to > > the VM setup? > > Define 'any'? If you're idea of virtualization is to oversubscribe > servers and hurt performance, then no. > > To mix audio, the code takes lots of audio slices and merges them with > an algorithm. But if the underlying cpu doesn't provide consistent, > reliable ticks, as potentially happens in virtualization, then good > luck with what's going to happen to your audio mixing. > The MeetMe dependency on DAHDI is for the audio mixing. ConfBridge has no dependency on DAHDI for anything. -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!
- Original Message - > From: "David Backeberg" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, April 25, 2011 9:27:19 AM > Subject: Re: [asterisk-users] The new ConfBridge application is now in > Asterisk Trunk! > On Mon, Apr 25, 2011 at 9:38 AM, David Vossel > wrote: > > I am proud to announce that after a good bit of development, > > community feedback, testing, and >code review, the brand new > > ConfBridge application has been officially merged into Asterisk > > >Trunk!!! > > http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 > > > > If you are already familiar with ConfBridge from Asterisk 1.6.X and > > 1.8, forget everything you >know. This is a completely revamped, > > highly optimized, and feature rich conferencing >application capable > > of mixing sample rates from 8khz all the way up to 192khz! Exciting > > right?! > > So way back when the 'old' ConfBridge was announced, my understanding > was it was originally an internal Digium tool for exercising the > Bridge() code and it was decided to release it to the public in the > event the code might be useful to others. The old ConfBridge was > missing stuff that was in MeetMe(), and wasn't that compelling for my > particular usage. > > This 'new' ConfBridge looks to be much more full-featured. So can > anybody explain the motivation for this? Is this a replacement for > MeetMe() where at a certain point we envision dropping MeetMe() from > the codebase? We needed a next generation conferencing application that could handle dynamic sample rates. Meetme's mixing required the use of Dahdi and was locked in at 8khz. This prompted the discussion of creating a new conferencing application to remove the Dahdi dependency and handle internal mixing of all possible sample rates. Since a re-write was necessary to achieve this, a new configuration method was designed that we believe is much more powerful than MeetMe's configuration method. There is no talk of removing support for MeetMe right now. We know people depend on MeetMe, so rest assured it is not going anywhere any time soon. > Does ConfBridge() scale to many users as nicely as MeetMe? I'm > assuming the MeetMe ability to use a hardware source for timing will > still be superior with large user counts in rooms? Incorrect, ConfBridge scales better than MeetMe. Not just a little better, but a lot better. My none official internal tests showed ConfBridge to be capable of 2 to 3 times more concurrent users than MeetMe using the 'drop_silence' confbridge.conf option. -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!
- Original Message - > From: "C. Savinovich" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, April 25, 2011 9:40:06 AM > Subject: Re: [asterisk-users] The new ConfBridge application is now in > Asterisk Trunk! > Does this ConfBridge requires a hardware timing source? Will I be able > to use this on any virtual server without having the need special > changes to the VM setup? Correct, dahdi timing is not required. If you can use the timerfd, or pthread timing modules you can use ConfBridge. -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!
- Original Message - > From: "Richard Kenner" > To: asterisk-users@lists.digium.com > Sent: Monday, April 25, 2011 8:42:07 AM > Subject: Re: [asterisk-users] The new ConfBridge application is now in > Asterisk Trunk! > > To help get you started, Malcolm Davenport has written some > > fantastic > > documentation on the asterisk.org wiki. It can be found below. > > I've looked at this documentation, but can't find the documentation > the > realtime interface, which is needed in order to schedule conferences > in > the future. Is the documentation missing or is this not part of the > application? We depend on it HEAVILY. > No, conference scheduling is not a feature that we have built directly into ConfBridge, and I'm debating on what it would look like. This is primarily a result of taking a different approach to the way the conferencing application is configured. Conferences in confbridge are only created dynamically. The scheduling would probably have to live on the user or bridge profile. So a dynamic profile would be created and that profile would have a start and end time. I'll have to think about this more. While it isn't simple, there are existing tools that can be leveraged to achieve similar functionality. For example, func_curl can be used to pull this conference information from a database directly in the dialplan, or an AGI script can be used to determine when a conference is available or not. The hard part would be warning the conference users the conference is about to end. This would require some dialplan cleverness. I suppose joining a local channel into the conference using the Wait and Playback application would work. The local channel could be the marked user, when it enters the conference begins, and when it leaves everyone is kicked. The local channel would just wait for the duration of the conference, and can provide playback warnings towards the end. I hope this helps, If you have any feedback on how you would like this feature to look in with the new ConfBridge configuration system please let us know. If the feature is well understood enough, perhaps someone will come around and develop it before Asterisk 1.10 is released. We are marking this as a feature request on the wiki, Thanks! -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The new ConfBridge application is now in Asterisk Trunk!
Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely revamped, highly optimized, and feature rich conferencing application capable of mixing sample rates from 8khz all the way up to 192khz! Exciting right?! So Go! use it, test it, and report back! Tell us what you like, what you don't like, if you want a feature that doesn't yet exist, and report bugs. This conferencing application has huge potential and we need community feedback. Asterisk 1.10 isn't that far away, and once it is branched adding new functionality to this application may not be possible, so start using it now! To help get you started, Malcolm Davenport has written some fantastic documentation on the asterisk.org wiki. It can be found below. https://wiki.asterisk.org/wiki/display/AST/ConfBridge+1.10 New conference join and leave sounds have been created for this application, but will not be available officially until the next sounds release. If you can not wait until then you can find them attached to this issue, https://issues.asterisk.org/view.php?id=19165. If you start using the ConfBridge application and find that you are interested in writing a new feature for it, feel free to use me as a resource by email or IRC. I'm happy to review your code and anything else I can to do make this application successful. Thanks! -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
