similar to: random sections lost from call recording

I've compiled ztdummy, following the directions on http://www.voip-info.org/wiki-Asterisk+timer+ztdummy, and when I try to modprobe zt : #modprobe ztdummy FATAL: Error inserting ztdummy (/lib/modules/2.6.18-gentoo-r6/misc/ztdummy.ko): Input/output error FATAL: Error running install command for ztdummy I haven't been able to find any info on what the I/O error might be or how to

Hi, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

Hi I've got a script like this exten => s,1,SetVar(CALLFILENAME=/var/www/recordings/${TIMESTAMP:0:8:7}/${UNIQUEID}) exten => s,2,AGI(recordstart.py,${ARG1},${CALLERIDNUM},${CALLFILENAME},Ind) exten => s,3,DIAL(ZAP/g2/${ARG1},70) exten => s,4,AGI(logerror.py,${ARG1},${CALLERIDNUM},${CHANNEL},${DIALSTATUS},${DATETIME}, ${CAUSECODE}) exten => s,5,hangup exten =>

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

I don?t know if the last message was with content. So, I sent again. I have installed a Digium card TE210P and unicall for use MFC/R2. I think that it?s all right but I can?t make and receive calls. I?m using asterisk 2.1 with the patch made by Jos? P. Leit?o and the follow libs: libsupertone-0.0.2 libunicall-0.0.3 libmfcr2-0.0.3 zaptel 2.1 My number is 34318300. The Telco send me only 8300.

Dear All I recently proposed a simple modification to Wedderburn's 1974 estimate of overdispersion for count and binomial data, which is used in glm for the quasipoisson and quasibinomial families (see the reference below). Although my motivation for the modification arose from considering sparse data, it will be almost identical to Wedderburn's estimate when the data are not sparse.

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination

I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

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, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [root@redhat monitor]# sox in.wav in-rev.wav reverse [root@redhat monitor]# [macro-record-cleanup]

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Extensions No Problems Panasonic Ext -> SIP Extensions No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1]

I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect). When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Ext No Problems Panasonic Ext -> SIP Ext No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",

[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

I currently have my Asterisk set up to "monitor" (record) all audio in my conference room on meetme. However, Asterisk will record an "____in.wav" and "_____out.wav" file for each user that joins the conference. Is there a way to set my extensions.conf file up so it only records when user when extension 1234 calls, for example? I'm assuming that the

1. RE: Start recording automatically when xferring to anextension? (Don Pobanz) Message: 1 Date: Fri, 25 May 2007 11:54:33 -0500 From: "Don Pobanz" <dpobanz@hastingsutilities.com> Subject: RE: [asterisk-users] Start recording automatically when xferring to anextension? To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Hi, I like my recordings to go to date wise folder i mean to say that for example today is 20070409 so all recordings should go directly to that folder instead of one folder for whole month. and then next day's recordings should go to next date folder. so how can i do that my current monitor context is like... exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb)