similar to: soxmix

I've tried to simplified the dial plan and use "n" instead of numbers but I've noticed it is not executing my voicemail if I substitute number with "n" In the example below when the call is not answered, it does not go to voicemail; call just hangup. exten => 1,1,Playback(transfer) exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw) exten =>

I'm trying to to to "dial1" if caller id match: but dial plan execute 220,n(dial1) regardless exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1) exten => 220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w) exten => 220,n, I was under impression that if condition is met it will jump to

Asterisk Monitor seems to be working fine. Though the problem I am having is the two files (in & out) muxing. I added ,m to the string, yet the call records two files still, and I get the resulting error, at the bottom. monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4 8:23-in.gsm

Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix on my linux shell it works so i can mixed two calls into one consolidated call. I want to do the process automatically since extensions.conf but it doesnt work. My extensions.conf looks like this: exten => 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)

Hi, is there anybody who knows how to force Asterisk 1.4 to use soxmix instead of sox? Thank you. Giorgio

Hi, Monitor and soxmix (m option) work fine in CVS Head, not in Asterisk 1.0.9, as the Wiki says. http://www.voip-info.org/tiki-index.php?page=Monitor+setup+sample Anyway I am wondering why asterisk 1.0.9 console shows on Hang up: "monitor executing ( nice -n 19 soxmix "//var/spool/asterisk/monitor/45/47-20050812-113631-in.wav"

In asterisk 1.6 the voicemail prefix u b don't work, I have: exten => 1,3,Voicemail(u11) and it keeps telling me: No entry in Voicemail config file for "u11" exten => 1,3,Voicemail(11) works, Isn't prefix "u" suppose to play: "The person at extension ... 11 ... is unavailable," ? -- Joseph

Is the path to soxmix in the $PATH environment variable when asterisk starts. If you're running from an init script it may not have path set at that point. When you log in, you set the path variable. Have you tried putting explicit paths into the command in your extensions.conf? IE /usr/bin/soxmix instead of just soxmix. HTH Chris That sorted it, thanks

Is anybody using the Copula package in R? The particular problem I'm facing is that R is not acknowledging the fitCopula command/function when I load the package and (try to) run something very simple: fit1 <- fitCopula(x1 = list(u11,u12,u13,u14,u15,u16,u17,u18), tCopula, optim.control = list(NULL), method = "BFGS") Anybody also using it, successfully or unsuccessfully?

I need a client for Solaris 10 u11 that will allow me to mount encrypted shares located on a EMC VNXe3200 Thank you for your time ! Charles Corpany Charles.Corpany at ngc.com<mailto:Charles.Corpany at ngc.com> Information Systems Sector, Civil Division - VITA Program Northrop Grumman Corporation Virginia Department of Health Datacenter James Madison Bld, Upper Basement, UB20E 109

I have Asterisk configured to record calls. Both in and out record ok but SoxMix fails to join the two files. The error from the CLI is as follows: Execute of ( nice -n 19 soxmix /var/spool/asterisk/monitor/Support-in.wav /var/spool/asterisk/monitor/Support-out.wav /var/spool/asterisk/monitor/Support.wav && rm -f /var/spool/asterisk/monitor/Support-* ) & failed. If I run exactly the

Hello, I have the following error when checking for GPOs for a single user, listing all GPOs is working: ------------------ root at tn2-debian1:~# samba-tool gpo listall GPO : {31B2F340-016D-11D2-945F-00C04FB984F9} display name : Default Domain Policy path : \\example2.net\sysvol\example2.net\Policies\{31B2F340-016D-11D2-945F-00C04FB984F9} dn :

Hello, My company is doing some * testing with our Class 5 softswitch and had some questions regarding ringback being provided to our PSTN users (off --> on net calling) Currently with MGCP subscribers, we know the PSTN ringing is provided by a digital PBX for example, However, it looks like with SIP, our softswitch is relying on MGCP signaling on our PSTN gateways to provide ringback

Hi All: I encountered a problem with smbmount and win2003 sp1. Before I upgrade sp1, everything works fine. But after sp1, smbmount = can't work. The command and error message are as follows: smbmount //localhost/share1 /tmp/mountest -o = username=3Ddomainu1,password=3Dpassword ERRDOS - ERRnoaccess (Access Denied.) SMB connection failed I also checked the release note about the issue, and

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:

Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix

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

When I use sox-12.17.5 recording and mixing works fine but removing the -in.wav and -out.wav file doesn't work. When I tried sox-12.17.6 recording doesn't work but removing the IN / OUT wav files is working. Anybody has a similar experience. The command didn't change but it seems to me there is a timing error: The creation time for the file is: Jan 29 15:30 18-20050129-152954-in.wav

Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints?

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in