similar to: ANNOUCEMENT: Tool to combine call recording into a single 'stereo' file

Displaying 20 results from an estimated 2000 matches similar to: "ANNOUCEMENT: Tool to combine call recording into a single 'stereo' file"

2007 Nov 20
1
[asterisk-dev] trunk working under windows!
Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their first steps in the asterisk world Crossposted to -users. Zoa Luigi Rizzo wrote: > As a
2005 Feb 16
2
Monitor does not like variable subsitutions
Hello, I have been attempting to get the Monitor function to accept a loal variable substitution in order to use the same filename later in the same context. Monitor does not appear to like it, as it attempts to use wav|filename as the recording type, as opposed to just wav. Here is what I get if I just supply a filename directly (it works fine): --context----------------------------- exten
2005 Aug 14
1
ogg causing me heart burn
Dear forum, I have a install of asterisk using AMP. I followed the install guide off the AMP site. http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf When I start using amportal start or asterisk -ccccv I received this in my log. The last line is that ogg failed. I have found nothing on the web about this, and I am not even sure where to start troubleshooting. Any help
2005 Jun 23
0
Voicemail recording cutoff when silent for 1 second
I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as
2005 May 20
0
Annoucement in MeetMe and segmentation fault
Hi! I have MeetMe working fine except I thought that if I used the "i" option, there would be some kind of announcement made upon a new meetme person joining a conference. Is that correct? Is it the "i" option? Because I'm not hearing an annoucement when a new person joins the conference. Also, if I combine the "i" with the "x" option, there is
2015 Apr 01
0
help : annoucement queue
Hi everybody, I've a matter with the queue annoucement with the "thereare", because if I put just one member in my configuration (member => SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add one member (eg: member => SIP/2098 and member => SIP/2099), the ivr don't gave me the range but It play the
2006 May 03
0
ANNOUCEMENT - LUSTRE ROLL for Rocks 4.1
Dear all Scalable Systems is pleased to annouce the availability of the Lustre Roll for Rocks 4.1 (Preview Release). The Lustre Roll contains the latest Lustre 1.4.6 version software packaged to work with Rocks 4.1. This is a *PREVIEW* release based on V1.4.6. The final and production release which we will support officially will be based on V1.6. Note that the Lustre Roll will install Lustre
2005 Sep 08
0
Icecast 2.3.0 RC2 Annoucement
The Icecast development team is pleased to announce RC2 of Icecast 2.3. This will most likely be the final RC before the 2.3 release. We appreciate everyone who has tested the RC's so far. All bugs (for this release and all others) should be logged in Trac - http://trac.xiph.org. The downloads: Tarball : http://downloads.xiph.org/releases/icecast/icecast-2.3.0.rc2.tar.gz SRPM :
2005 Sep 08
0
Icecast 2.3.0 RC2 Annoucement
The Icecast development team is pleased to announce RC2 of Icecast 2.3. This will most likely be the final RC before the 2.3 release. We appreciate everyone who has tested the RC's so far. All bugs (for this release and all others) should be logged in Trac - http://trac.xiph.org. The downloads: Tarball : http://downloads.xiph.org/releases/icecast/icecast-2.3.0.rc2.tar.gz SRPM :
2005 Sep 23
3
Icecast 2.3.0 Release Annoucement
We are pleased to announce the next release of Icecast. Downloads available at http://www.iceast.org/download.php (pending mirror propagation) **** New features for 2.3.0 **** - Streaming support for ogg speex, ogg flac, ogg midi - intro file support - per mount settable Intro files will play when a listener first connects to a stream. This is designed for station jingles and the
2005 Sep 23
3
Icecast 2.3.0 Release Annoucement
We are pleased to announce the next release of Icecast. Downloads available at http://www.iceast.org/download.php (pending mirror propagation) **** New features for 2.3.0 **** - Streaming support for ogg speex, ogg flac, ogg midi - intro file support - per mount settable Intro files will play when a listener first connects to a stream. This is designed for station jingles and the
2005 Sep 12
0
Icecast 2.3.0 RC2 Annoucement
Ok, this bug has been identified, logged, and fixed in SVN. We will be most likely building an RC3 in short order... oddsock At 03:43 PM 9/9/2005, Joel Ebel wrote: >Playing around with it some, it seems to be related to the web >interface. If I don't touch the web interface it seems to stay running, >but when I start trying to view pages in the web interface it quickly dies.
2000 Apr 18
0
annoucement for stochastic processes workshop
Hello all, I wanted to share with you this announcement for an upcoming workshop on the analysis of neural data to be held this summer at woods hole, MA. The workshop has been extremely productive in the past years and we anticipate another good working group this year. This workshop is applicable for any working in time series analysis whether it involves point processes or sampled continuous
2015 Mar 31
0
help : annoucement queue
Hi everybody, I've a matter with the queue annoucement with the "thereare", because if I put just one member in my configuration (member => SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add one member (eg: member => SIP/2098 and member => SIP/2099), the ivr don't gave me the range but It play the
2008 Dec 04
6
OpenGL test?
First off I just want to say thank you to all who contribute to wine, it is an awesome piece of software for the Linux community. I am trying to find a way to do a simple OpenGL test within Wine. I have looked all over but I can't seem to find anything. What is the easiest way to test OpenGL and ensure that it's working correctly? The reason I'm asking is this: I recently upgraded
2008 Dec 24
0
[Annoucement] Compiz feature branch compiz++
Hi, I've currently pushed a new branch called "compiz++" to the freedesktop repository, with some features I've been working on during last months. Because most of the features also require (BIG) changes to the plugins, I've decided to put them all together. - No direct access to member variables: Everything is now done with getter and setter functions. This helps with the
2016 Mar 04
2
Asterisk 13.5 and higher (asterisk 13.7.2) quitting
I start asterisk 13.7.2 and it dies before I can rasterisk into it. I've tried getting a coredump, but it doesn't coredump. I know there are a lot of errors in the log below, but most of those just say it'll not load a module, and no big deal. When launching from commandline (not service script) here is what happens. http://pastebin.com/3GFe6fG9 Travis Ryan Director of
2005 Oct 03
1
suse 9.3 pro asterisk install from source problem
Hi, Can any one help I'm trying to install asterisk on suse 9.3 pro from cvs release v1_0 version 1.0.9 and when I try to make from the asterisk directory I get the following error. Is there anybody that could give me a pointer as to what the issue may be? DDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN
2007 Jul 09
10
Monitor events?
Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel
2008 Jun 30
4
Voicemail- Recorded Mesage Low Volume
> Hi Daniel, > > I'm intrigued by this and wanted to try it out - but I'm wondering how > you get Asterisk to call sox at all during Voicemail()? Our server > doesn't even have sox installed, so I'm not sure how to go about > tricking Asterisk into running a different one. To do anything useful you would have to get sox installed on your server. But to get