similar to: Solved? => Playback audio before answered by a queue member

On Thu, Sep 24, 2020 at 05:32:31PM +0300, Eli Cohen wrote: > Change core vdpa functionality into a loadbale module such that upcoming > block implementation will be able to use it. > > Signed-off-by: Eli Cohen <elic at nvidia.com> Why don't we merge this patch together with the block module? > --- > V0 --> V1: > Removed "default n" for configu options

On Tue, Sep 29, 2020 at 09:20:26AM +0300, Eli Cohen wrote: > On Mon, Sep 28, 2020 at 03:55:09PM -0400, Michael S. Tsirkin wrote: > > On Thu, Sep 24, 2020 at 05:32:31PM +0300, Eli Cohen wrote: > > > Change core vdpa functionality into a loadbale module such that upcoming > > > block implementation will be able to use it. > > > > > > Signed-off-by: Eli

On Tue, Sep 29, 2020 at 09:34:33AM +0300, Eli Cohen wrote: > On Tue, Sep 29, 2020 at 02:26:44AM -0400, Michael S. Tsirkin wrote: > > On Tue, Sep 29, 2020 at 09:20:26AM +0300, Eli Cohen wrote: > > > On Mon, Sep 28, 2020 at 03:55:09PM -0400, Michael S. Tsirkin wrote: > > > > On Thu, Sep 24, 2020 at 05:32:31PM +0300, Eli Cohen wrote: > > > > > Change core

On Tue, Sep 29, 2020 at 09:57:44AM +0300, Eli Cohen wrote: > On Tue, Sep 29, 2020 at 02:51:12AM -0400, Michael S. Tsirkin wrote: > > On Tue, Sep 29, 2020 at 09:34:33AM +0300, Eli Cohen wrote: > > > On Tue, Sep 29, 2020 at 02:26:44AM -0400, Michael S. Tsirkin wrote: > > > > On Tue, Sep 29, 2020 at 09:20:26AM +0300, Eli Cohen wrote: > > > > > On Mon, Sep

I have a WSDL and used to wsdl2ruby to create the default.rb and defaultdriver.rb ****************************************************************** the method that i''m calling class representation is: # {http://home.setup/abs}DeliverMessage class DeliverMessage @@schema_type = "DeliverMessage" @@schema_ns = "http://home.setup/abs" @@schema_qualified =

We would like to reserve several FLAC Application ID's: App Name: Myriad App ID: "AFMY" / 41454d59 URL: http://www.audiofile-engineering.com/myriad App Name: Triumph App ID: "AETR" / 41455452 URL: http://www.audiofile-engineering.com/triumph App Name: Fidelia App ID: "AETW" / 41455457 URL: http://www.audiofile-engineering.com/fidelia Contact Info: Matthew

--- Brady Patterson <brady@spaceship.com> wrote: > > Brady, I would say for now, your proposal is fine. I am going > > to move flac to getopt soon... > > > > Matt, that would be cool if you wanted to take on the audiofile > > support. I have actually been waiting to ask you for that, > > waiting until I got the getopt support done, and the new > >

Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI> queue show 0317998989 0317998989 has 0 calls (max unlimited) in

On Jan 3, 2007, at 3:28 PM, Josh Coalson wrote: > the FLAC parts look OK but I don't know how the audio converter > works. I think that's where the problem is. it is probably > converting float to 32-bit int full scale. well, here's what I found out about the 32-bit integer conversion. If the original 16-bit sample is 0x52F3, it goes through floating point, then comes

I'm in search of useful ACD type statistics from the queues. Ie talk time, ratio's, dropped calls etc. The flat file queue_log is nice, but more useful would be the data in Postgres or Mysql. Unfortunately the queue module does not yet support ODBC DB logging (yet). In the meantime this quick and dirty hack gets the job done. Replace the flat file with a unix named pipe.

Hi All, I've got a realtime queue in place (strategy is "wrandom"), and have added a member dynamically via "queue add member ". My agent shows in the queue, but when he gets the call is not recognized as "In Use". Here is the output from "queue show" prior to the call: *CLI> queue show QUEUE_3 has 0 calls (max unlimited) in 'wrandom'

I am trying to insert a bit of ogg vorbis decoding into my MFC/C++ program and send it to my waveOut functions. // AudioFile.cpp #include "vorbis/codec.h" #include "vorbis/vorbisfile.h" . . . OggVorbis_File vf; FILE* pFile; pFile = fopen("verdes_2.ogg", "rb"); int err = err = ov_open(pFile, &vf, NULL, 0); . . . ov_clear(&vf); In my

Hi, When I build the spandsp library needed for the FAX application, running ./config I get the following with 'no': [root@d17 spandsp-20031021]# ./configure ... checking for _doprnt... no .... checking for pow... no checking for sqrt... no checking for rint... no .... checking socket.h usability... no checking socket.h presence... no checking for socket.h... no .... checking audiofile.h

On Tue, 30 Jul 2002, Josh Coalson wrote: (Matt wrote): > > A more flexible approach might be to use libaudiofile or a similar facility, > > so that multiple input formats can be automatically recognized, and future > > formats supported in a generalized way. Agree completely. I did this because I wanted to get something working with as small a change as possible. > Brady, I

Hello - I am upgrading from asterisk v1.2 to v1.6 and I am seeing a problem with recording CDRs using MySQL. Unlike all of the other postings and web pages I have found on this issue, my installation successfully stores the -first- CDR, but nothing after that. As background info, I will note that I don't use CDRs for billing, but more in a logging fashion, to record how a given call

--- Matt Zimmerman <mdz@debian.org> wrote: > On Tue, Jul 30, 2002 at 09:04:38PM -0500, Brady Patterson wrote: > > > The patch I submitted only reads AIFF files. I'm about to start > the patch to > > write AIFF files. > > > > To do so, we need a command-line option to specify AIFF. My > inclination is to > > add an option: > > > >

Hi All My dial plan has the following context: [sip-guest] exten => _!.,1, Answer exten => _!.,n, verbose(1,[${EXTEN}@${CONTEXT}]) exten => _!.,n, resetcdr(w) exten => _!.,n, resetcdr(w) exten => _!.,n, set(DNIS=${EXTEN}) exten => _!.,n, resetcdr(w) exten => _!.,n,

(Both on Asterisk 1.2 and 1.4) I was struggling to find out why my CDR was recording dst = h after a call hangup. It was working fine until I added a GotoIf statement before ResetCDR to calculate some value for userfield column. Today I tested and found out that if ResetCDR is put after GotoIf (or after if in AEL), it doesn't record correct value in dst column, and isntead puts 'h'

What would it do if you exten => h,1,ResetCDR(w) exten => h,2,NoCDR() exten => h,3,DEADAGI(get-unqiueid.php) I have not tried it but in theory it should write the first CDR and then kill the write of the second NO ANSWER CDR. Let me know if it works for you as I may need to do it on some of my h exten code as well. Bryant ---------------------------------------- From: