similar to: Monitor app

Displaying 20 results from an estimated 7000 matches similar to: "Monitor app"

2003 Jul 01
6
Enhanced queue app
To all who need more queue functionality, We are contracting Digium to enhance the queue app for our call center needs. Please read the following email conversation and give your ideas. Unless a glaring omission is found in my specification we will have them start tomorrow (Wednesday). I may not have thought of something important. It will be released to all Asterisk users by Digium. Thanks for
2003 Jun 18
2
Wrap-up
Is it possible to specify a 'wrap-up' time in a queue so agents will have a specified amount of time to complete tasks between calls unless they hit a key on the phone? As it is they can recieve a call moments after they hang up with no 'down time'. Thanks Jim Friedeck
2003 Aug 08
3
segfaults with queue
Just cvs'ed about 40 minutes ago (10:15 CST 8-8-03). Segfaults when I use a queue app in many different scenarios. When calling phone is only member of queue I get a segfault. When 1st called extension is outside line I get a segfault. Many other scenarios as well. Unsure how to go about troubleshooting. Any ideas? Jim Friedeck
2003 May 12
1
Gastman compile errors
I seem to be encountering a problem with the db.h file when compiling gastman. Incorrect definitions for functions and so on. Here are the errors: If I don't copy the db.h from asterisk source to the include directory of gastman I get: gui.c:31:16: db.h: No such file or directory If I do then I get: gui.c:743: too many arguments to function This is for the line: if (!(res =
2003 Aug 06
10
AgentCallbackLogin
I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: queues.conf: [general] [default] [q_lo_1] music = default strategy = ringall context = c_in_1 timeout = 15 retry = 2 maxlen = 0 member => Agent/@3 agents.conf: [agents] autologoff=10 wrapuptime=15000 group=1 agent => 1001,1234,Agent1 agent =>
2003 May 19
1
Call Group
Anyone know how to specify a call group (as specified in the example zapata.conf) when using Dial? No other reference I can find. Jim Friedeck
2003 Aug 08
3
Killing runaway PBX
How do I stop asterisk when it is in a bad mood? It keeps dialing extensions and won't listen! I tried kill <PID>. No go. I don't want to have to reboot again. Thanks. Jim Friedeck P.S. I love it when my boss looks over my shoulder and I don't have an answer when he says: 'So, what are you doing?'
2003 Aug 11
3
Ring while on phone
Our CSR people need to be informed when a call is ringing in when they are on the phone. Is there a mechanism for informing an off-hook target channel of an incoming call? We have a guy who should get first shot at all incoming calls on our local lines and our customer service line. If he is on the phone, he should get beeped and then be able to place the current call on hold to answer the
2003 Nov 02
3
recording files for menues
How do you suggest doing that? How can I convert wav files to gsm files? thanks Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: shoval@softov.co.il Mobile: 972-55-229220 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031102/f84d7805/attachment.htm
2003 May 23
1
Channel Status in AGI
I am looking for a way to quickly and easily test for on-hook channels from within a C-language AGI app. CHANNEL STATUS works but is a bit clumsy. I don't want to rely on strcmp'ing the returned '200 result=-1' as the meaning of this might change in the future. I am trying to create an ACD using MySQL and want to test each channel before I Dial it. Any ideas? Jim Friedeck
2003 Jun 12
3
Monitor application
Hi, I've had a search through the archives and didn't find much. Is anyone using the Monitor application? I have it working but there is a really big drawback. The files are always called the same thing, which means if I make 2 calls one after the other the first recording is lost. I half expected Monitor to use something like ZAP-2-1-<yyyymmddhhmmss>-in/out.wav for it's
2005 Sep 18
5
Monitor and sox mix quality
Hello All, I am using monitor with soxmix, however the quality seems somewhat low after sox converts to mp3. Does anyone know a way to get a higher quality file? Some of my lines are coming in on isdn. Regards, Greg
2004 Apr 23
3
MP3 encoding of Monitor files
I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call:
2006 Mar 14
3
Voice volume using Monitor application
I am using the Monitor() application (with soxmix for combining the audios) and the voice connected to the phone network is recorded at a lower volume then the voice connected directory to the Zap analog phone card. How can I get both the audios to be at the same volume on recording? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL:
2006 Feb 10
1
2wav2mp3, monitor, mixmonitor, mpg123, queues
Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten
2008 Apr 21
2
Monitor not merging calls
I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>
2005 Feb 15
7
Extra sounds (Weather)
Does anyone know of a AGI script that takes advantage of the weather sound files that's included with the extra sound files available from www.loligo.com/asterisk/sounds/ <http://www.loligo.com/asterisk/sounds/> ? Thank, Jeramie -------------- next part -------------- An HTML attachment was scrubbed... URL:
2003 Oct 07
5
IAX and Jitter problem
Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been
2006 Oct 16
1
Monitor stops recording midstream?
Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 running Linux on 2006-06-17 When I used monitor, I seem to get most calls cut off if they run very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any ideas what might kill the recording process? I'm beginning to wonder if soxmix is truncating the file when it blends the in/outbound streams together
2009 Dec 15
2
monitor-type=MixMonitor
Hi! Since we upgraded to 1.6.1.11, asterisk is only outputing monitored files -in and -out. It is not mixing them in the end. queues.conf has monitor-type=MixMonitor... Would somebody help me debug why it doesn't mix the sounds?? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: