similar to: After Monitor() files disappear

Hi there, I searched in mailinglist and in web, but no answer to my problem... Only this post with no answers: http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple controller support). In my Asterisk-box there are 2 Fritzcards (module for second card compiled with changes on sourcecode found in the web).

Anyone here running asterisk on openvz, if so what are your experiences? Right now we are trying to tune out the resources for the difference VEs, but not with a whole lot of luck. Just wondering if someone watching could shed some like on what has worked for them, and how many exts/simultaneous calls etc are happening. Thanks Miles -------------- next part -------------- An HTML attachment was

Hi all ! I'm trying to make a automonitor generated filename to "make its way" into CRD(usrefiled), so I can keep track of recorded conversations in CDR logs. Looking how to do that, I have found cool (but almost undocumented) option of res_monitor: if you set monitor format in form of "format:<string>" (i.e. "wav:monitor"), res_monitor will prefix the

Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends "Remote hold" message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hello, I have a little problem with signalling. An E100p is connected to an Alcatel PBX, wich has an E1 to the outside. Located in Germany. zapata.conf: switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes signalling=pri_cpe .... With asterisk 1.0.2 I can call from a SIP phone to a phone connected to the Alcatel and the SIP number is correctly displayed at the caller.

Hi, on AIX, mkstemp doesn't open a file with the O_LARGEFILE option, so you can't transfer files > 2GB to an AIX machine. Here is a fix: diff -c -r rsync-2.5.6.orig/syscall.c rsync-2.5.6/syscall.c *** rsync-2.5.6.orig/syscall.c Sun Jan 26 21:09:02 2003 --- rsync-2.5.6/syscall.c Wed May 14 13:55:15 2003 *************** *** 151,157 **** if (dry_run) return -1; if

Hi, I have been making an RPM of FLAC to bundle with GStreamer. In order to get it working I had to make some rather hackish solutions in the SPEC file. The flac Makefile does to build into the correct directories while creating an RPM for some reason. I have attached the SPEC file I ended up with if it is of interest. Of course it didn't help me much cause it turned up we had a bug in the

Hello list, I need to record all calls. So I'm using application Monitor. Works good until someone transfers a callee to another internal extension. Example: A calls B A set B on hold A calls C A transfers B to C with SIP transfer (SIP REFER - with phone funktions and not Asterisk attended transfer). I found http://bugs.digium.com/view.php?id=0013538 . "corruptor" asked about this

Hello, I want to enable call forwarding at my telco. In Germany you can press *21*destination# and all calls will be redirected to the destination without interaction with any equipment on my side. How to dial this with Asterisk and Zap-Channels? It can not be send as "called number", it has to be send as "keypad facility". Anyone here with some hints? The application

Hey Matt I wonder if it is possible that it doesn't work with AEL, does this seem ok to you? s => { Ringing(); wait(2); Answer(); Set(MONITOR_EXEC=/etc/asterisk/lameconvert.php /var/spool/asterisk/monitor/^{MONITOR_FILENAME}); Queue(MyTestQ,ni,,,18); Hangup(); } I have debug

Hi all Does anybody know where this command is supposed to go? Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MONITOR_FILENAME} /tmp/^{MONITOR_FILENAME}) In the queues.conf file it talks about it. So I naturally thought after I set up my monitor with monitor-format = wav monitor-type = MixMonitor That I could put a lame command in there to convert and move the file elsewhere for backup with

Hello, I have a problem with a Grandstream being behind a symmetric nat. The box which does the nat is a german "Fritz Box". This one does nat for the internal network. In the internal network is a Granstream BudgeTone 100. The nat router has a dial-up connection, so ip changes on every dial-in. |------------| |------------| |--------| |Grandstream

Hi, Just wanted to let you know that we have just made our GPL toolkit "Gemeinschaft" available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see

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

If you execute cat /dev/urandom at the xm console of a guest domain, it will spew garbage forever. Attempts to run xm destroy on it simply hang. _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

TOUCH_MONITOR is the variable to set if I need to specify my own options for 'One Touch Record' (filetype|filename|m). I cannot get it to work. Can you help? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050219/29dac9ed/attachment.htm

I want to set up a conference call to be recorded automatically, so I'd like the recording to start when the second caller joins the conference (one caller already there). The recording would continue until the last user hangs up. How can you determine how many are already in the conference bridge? [conferences] exten => 66,1,Answer exten => 66,n,Wait(1) exten =>

Was using this monitor line to get soxmix to mix test-in.wav and test- out.wav into test.wav. exten => 1200,1,Monitor(wav|/tmp/test|m) When I start the conference, the * console shows this: monitor executing ( nice -n 19 soxmix "//tmp/test-in.wav" "//tmp/test- out.wav" "//tmp/test.wav" && rm -f "//tmp/test-"* ) & /tmp shows test-in.wav,

Hi all, can anybody tell me how I get asterisk calling an executable after a queue call? only setting MONITOR_EXEC and MONITOR_FILENAME does not work anymore! We use normal monitor to record _in and _out files. does work perfectly. But calling an exec does not do anything. We also tried setting MONITOR_OPTIONS=b Can anybody tell us, how we can get that running? We would like to call