similar to: RE: [Asterisk-Dev] Several patches, includin g recording and music -on-hold

Mahmut, excellent summary :-). I look forward to your next update. One little thing, In the manager events that show start/stop monitoring, can you please include a field that indicates the filename(s) to which the monitoring was written? Thanks, Ben -----Original Message----- From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au] Sent: Tuesday, April 15, 2003 5:17 AM To:

Mahmut, First of all, I'd like to reiterate what a great patch this has been. I'd also like to voice support for having an option to mix the files on the fly, and name them uniquely. While I was able to smoothly put the files together with soxmix, I see on the fly mixing as hugely beneficial to an automated solution to saving/delivering the messages without intervention. One feature,

Hi Wade, Sorry for replying so late. I had been sucked into other tasks for a while and only now can catch up with the list. > When I dial my iaxtel number from my extension on channel > Zap/15, I get two > files recorded in /var/spool/asterisk/monitor: > > Zap-15-1-in.wav and Zap-15-1-out.wav and they sound fine. > > When I dial again, it overwrites the same two files.

Hi all, I'm using asterisk in a production environment now and this afternoon I got reports complaining that it was not working. Looking at the asterisk console output, I saw it contains lots of error messages as printed below. Unfortunately it is not obvious from the logs as to what started all this. Just before the error messages start, everything seems to be working fine with no problems.

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

[apologies for no line wrap; config lines at bottom] I have mentioned on several threads here that the Monitor application doesn't do exactly what one would expect: the originating and answering legs of a call are unsynchronized by the duration of the interval that it takes for the answering leg to pick up the phone. This can be very distracting in a final mixed version of the file. Brian

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

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

When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this: Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]]]]]) options r: Send ringing

I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't seem to get it to work correct. Could someone point me to what I need to do? I have attached what I believe are the relevant parts. [globals] ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 ; uncomment this line if you are using Ogg Vorbis

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"

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

Hi, I'm having high load, choppy sound and slow responsives with an asterisk server (version 1.2.12.1) that make a peak of 90 channels (around 60 phones calling at max, isn't necessary to reach this peak to get the problem). All the traffic is SIP, with recording for every call. The server has: Intel(R) Xeon(TM) CPU 3.20GHz (with HyperThreading disabled for inestability) 4G RAM 2 DD SCSI

Hello, We are working on a new Asterisk installation and have run into some problems related to playing MusicOnHold for a caller when they have been placed on hold by an agent, that took the call from a queue. A. When pressing the HOLD button on SNOM 190 and Grandstream BudgeTone SIP phones, MusicOnHold works fine when making inbound or outbound direct calls by extension. Music starts to play

Hello everybody, I'm trying to understand how can I set the MoH class for parked calls. I set the "incoming" class for calls, and it works correctly. When I park the call, the music on hold is ok, but when I close the communication on the parking side, the parked call gets the default music on hold class. Can someone explain me what's going on? Thanks in advance, -- Dott.

Type 'mpg123' at the Linux CL. (no quotes) If the version is anything other than 59r, you problem is solved. Go to the Wiki and search for Music On Hold. Do the install of version 59r ONLY as described in the docs. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neil A. Hillard

Hi, I've recently installed Asterisk and have got the majority of it configured (what an excellent piece of software it is, too), but I'm having a couple of problems. The first one is with music on hold! I've downloaded and installed mpg123 as specified: ># whereis mpg123 >mpg123: /usr/local/bin/mpg123 It's the correct version: >#

I couldn't find any specific reference to this but maybe Im missing something completly... anyways, when trying to mix a few wav files together post-recording (the -in/-out files) using a pretty vanila soxmix line, I get the error: Done Mixing OUT115-20060215-150749-1139976460.7898-out.WAV..... Mixing OUT115-20060215-155022-1139979011.8787..... /usr/bin/soxmix: invalid wav gsm frame size: 1

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