similar to: How do I reload extensions included in a switch statement in extensions.conf?

Mikael Magnusson writes: > Package: libspeex1 > Version: 1.1.6-1 > Severity: wishlist > > Is there anything that keeps 1.1.6-1 in experimental from being uploaded to > unstable? Do you know when it will happen? > > I'm debianizing iaxclient, which needs speex version 1.1.x. Currently I'm > using 1.1.6-1 in experimental, which is working fine. >

Did you check it against the trunk in SVN? If it's not applied, and you can hook Jean-Marc up with an email address like yours, I'm sure he will get right on it. :) Tom Mihai Balea <mihai@hates.ms> wrote: > > Hi all, > > Did anything happen to this patch? > It seems to me that it fixes a valid issue, but I'm not an expert. > Anyways, I didn't see

On the extensions.conf explanation page is a mention of the "switch" statement and it refers one to the "connecting two * servers" page. The only mention of the switch statement there is brief and in the example. However, the example seems to have some errors in it. It shows a sample of what's in the extensions.conf file, but it clearly has sections which would be in the

Hi all I'm trying to understand what the switch => statement in the extension plan exactly does. Unfortunately it seams to be very badly documented. There's allmos noting about it in o'Reilly's Asterisk Book. Nothing in the shipped /doc, nothing on voip-info.org. Even with google I found ony very few infos. So can somebody explain? I got two Asterisk Servers running

Hi all, i need to know how to use the switch statement in extensions.conf to throw calls on a secondary asterisk server. I have been trying all day to be able to make it work but its not working. It displays the following error: pbx_find_extension: No such switch 'master:secret at 192.168.0.7' I have read the help material on voip-info for connecting 2 asterisk servers and followed the

So I have two asterisk servers, one acting as a frontend and another acting as a backend interface to a PRI. All I want is for the backend to send all calls from the DAHDI interface to the frontend. Seems like switch would do that by placing this on the backend server: switch => IAX2/frontend1/inbound_context 'frontend1' is already defined with proper username/secret combos in

I checked it against the latest code in the git repository and it wasn't there. Mihai PS: if JM wants a @hates.ms address, I could prolly hook him up. Especially if he throws in some VAD code that's not "a hack" :) On Nov 5, 2007, at 2:08 PM, Tom Grandgent wrote: > Did you check it against the trunk in SVN? > > If it's not applied, and you can hook Jean-Marc up

Hello, Is it possible to use the switch => statement in extensions.conf (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to point to a database and in the database use the include-statement ? In extconfig.conf I would have : extensions => mysql,asterisk,extensions_table In extensions.conf I would then have : [includecontext] switch => Realtime/includecontext at

Mikael Magnusson wrote: > Jean-Marc Valin wrote: > ... > > I think the amount of Windows crap is getting a bit ridiculous. It always was. I'm just surprised you took so long to realise it :-). > The autotools files can also be used to build Win32 libraries using > mingw32 (preferably cross-compiling on GNU/Linux). It's currently > possible to build static libs. To

Package: xen-utils-4.1 Version: 4.1.3-7 Severity: normal pygrub does not recognize the include statement in extlinux.conf. The default Debian installation of extlinux creates a stub extlinux.conf which includes linux.cfg, where all the linux image entries are. This means that even if bugs 697407 and 697409 are fixed, pygrub still can't find the right kernel on a default Debian installation

Jean-Marc Valin wrote: ... > I think the amount of Windows crap is getting a bit ridiculous. Does > anyone know of a solution to keep things manageable? I mean, the same > autotools files manage the build for Linux PCs, Blackfins and MacOS (and > others). Yet, we need one set of files for each MS compiler version > (good thing they don't practice "release early, release

Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI> reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found

Can anyone provide more information on switch or point me to where I can find more about it? The only I've been able to find on the wiki is: http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers and towards the bottom of (section Forwarding to another Asterisk): http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Some of the questions I have are: 1) If I have an

Hi list! How to send a call directly to voicemail recording? When I put this exten => 313,n,VoiceMail,u221 Or this exten => 313,n,VoiceMail,b221 In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? -- Tomislav Parcina tparcina#lama.hr

Thank you all for the comments and suggestions! I got the link from Garbor to work, but that is the old r-travis approach (using C). I tried the same approach with native R Travis build but unfortunately it did not work. So I contacted Jim Hester and he told me that they are now actively working with implementing multiple R versions in the native R builds. /M?ns 2016-01-18 15:17 GMT+01:00 Dirk

Hi! Im currently developing an R package to automatically give students feedback on programming assignments in R. I use the testthat package as an engine for the unit testing and do a wrapper to make the automatic marking easy for the students. One function (called mark_my_file() ) will mark the students lab file using testthat tests and a new (simple) reporter. The problem is that I need to do

There is a problem in speex_preprocess_ctl. Both speech_prob_start and speech_prob_continue are set to 327.67 for all input values except 0 which results in 0. This is in floating point mode. I think the included patch fixes the problem. Mikael -------------- next part -------------- Index: libspeex/preprocess.c =================================================================== ---

Hi! I'm developing R packages and use Travis CI for continous integration. When submitting to CRAN Im suggestet to test the package using the latest R-devel. I would like that all test where run using Travis. Is there anyone who knows if this is possible to run travis test using the latest r-devel? -- Regards M?ns ============================ M?ns Magnusson 070 - 588 97 15 mons.magnusson

Dear Martin, Thank you for the well-reasoned response. I realized I was rather late to make this suggestion for 4.0.0, changing a somewhat low-level function that can indeed affect packages. I was just reviewing some R user scripts that were using type.convert(), mainly on data frames. In all cases, people were passing as.is=TRUE, so I was reminded that I would not be the only user who would

Hi I am having a problem with extension matching in asterisk (I am using asterisk 1.6.0.6). Is there a difference between extensions matching in realtime architecture and extensions matching in extensions.conf file. For example when I have these two extensions: -- _0699[134]XXXXX -- _06[069]XXXXXXX that are in the database and number 0699123123 comes in asterisk will always choose exten