similar to: Waiting before executing System command

One thing I forgot to mention: I also cropped the registration and cut marks off the sides of the pages. If you want the uncropped version, get: http://www.alexburke.ca/asterisk-tfot-uncropped.pdf Sorry about the excessive noise, but I figured I should mention this. >Date: Mon, 20 Feb 2006 18:55:50 -0500 >To: asterisk-users@lists.digium.com >From: Alexander Burke

Hello, world! I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently this works. I've read that the Zaptel package won't work on anything other than Linux, since it's intended to hook into the Linux kernel in the form of a kernel module. This concerns me, since I've read that ztdummy, the

Hello, list! I'm hosting a mirror of the book "Asterisk: The Future Of Telephony" by O'Reilly Press, published under the Creative Commons license; I believe this license allows me to do this, but if I'm mistaken, please let me know. I've taken the liberty of fixing the page numbers so Acrobat is now aware of the correct number of each page, and shrinking the

Hello, list! I'd like to create an extension that points to an offsite location (a number on the PSTN), the purpose of which would be to see if that offsite location is still on a call forwarded to it by Asterisk. This way a receptionist could choose to transfer calls to a mobile phone only if it's finished with the last call the receptionist forwarded to it. If I configure a custom

Hello all, I really appreciate the replies I've gotten about this so far (especially the support for wanting to run it on Solaris!). The core issue seems to have been missed, though -- is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? This probably comes down to a few issues: - Is ztdummy (a component of Zaptel) *really*

Hello, list! I'm having some trouble with A@H 2.7(?), Asterisk 1.2.5, inasmuch as my custom extension is not continuing execution when the caller hangs up. (Please excuse the sterilized output.) Here's how it's supposed to go: exten => 2,8,Monitor(wav,${TIMESTAMP}) exten => 2,9,Dial(SIP/Provider/8005551212) exten => 2,10,Macro(record-cleanup) If the caller hangs up

Hello, list! After Googling and checking out the voip-info wiki, I haven't had much luck in locating a decent web-based voicemail system for Asterisk to check your VM while you're away from the office without using a phone. Can anyone make any recommendations for such packages/applications? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada

Just in case anyone here hadn't noticed, Cisco is apparently making 7940/7960 SIP 8.2 firmware freely downloadable by anyone: http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 username: anonymous password: your email address -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada

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]

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 =>

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

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 thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

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,

I have a question on how to handle the "h" routines. I have noticed that if the call is hung up by the side that originated the call, the "h" routine is not extendable via a macro, or at least I have been unable to do it. My tests have included only SIP->SIP calls. If the originating side hangs up first: The macro is called from "exten =>

Hi all, I'm trying to make a context that will monitor a call and when it's completed it would e-mail the wav to a specified mail adres. So I made a standard context that records a call, like this: exten => _*31*00[1-9].,1,Setvar(CALLFILENAME=CALL-${EXTEN:4}-$ {TIMESTAMP}) exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m}) exten =>

OK. I now have call recording working for both incoming and outgoing calls. Now I want to make those wavs into mp3. I could launch a script from cron that checks for new wavs and converts them. But that wouldn't be so elegant. Launching it from * on hangup would be nicer. How is it done? [outgoing] exten => _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten =>

I'm trying to set up basic monitoring for a specific extension (5004) to record all outgoing and incoming calls and save them as WAV files. I've set this in the extensions.conf file: exten => 5004,1,Answer exten => 5004,2,Wait,1 exten => 5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-${CA LLERIDNUM}) exten => 5004,4,Monitor,wav|${CALLFILENAME} But it

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"