similar to: Asterisk Web-Based Voicemail?

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, 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, 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, all! I'm having a problem with the following snippet that executes upon hangup: exten => h,n,Wait(5) exten => h,n,System(mv /some/file /some/other/dir/) Wait() doesn't want to seem to wait! So instead I tried: exten => h,n,System(sleep 5; mv /tmp/${CALLFILENAME} /var/spool/asterisk/outgoing/) This only executes sleep, not mv. How can I make it wait before moving the

Hi, I used to have a Grandstream phone connected to Asterisk a few months ago. Worked just great! Then today I do a new install, rather than an upgrade, and all of a sudden I cannot check voicemail with it. No problem calling or receiving call. It simply speeds through the vm greetings but I cannot hear them. If I check the same VM with an analog phone it works fine. So I wanted to check

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'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

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

Hello all. Has anyone managed to get SIPCALL.co.uk's service working with the [*] box? I've managed to register with other SIP providers but not SIPcall. The debug just show's [*] attempting to register. But receiving a 401 error everytime. Cheers Matt

Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': Found Sheriff*CLI> Disconnected from Asterisk server -- Dave Cotton <dcotton@linuxautrement.com>

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

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

How to change available options (behaving) during listening of voice mail? (They are unnecessarily complicated) For example, I don't want to press 3 (advanced options) and again 3 for envelope. I just want to play envelope. Also, when saving message, I do not want to choose folder, I want that message as default be saved in old messages. And, I don't want to press 6 for next message, I do

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'

Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Running slackware 9.1 with compiled kernel from source 2.6.5 running ok. I have 2 HFC-S chipbased Billion Bipac PCI ISDN BRI cards installed in PC. Would like to use one card as in TE and one in NT mode. System works fine running pbx4linux.But want to use SIP functionality, so I would like to try out the Asterisk. Trying to install the bri-stuff.0.0.2.tar.gz (May 10 2004)package, getting the

Just idly watching * in console mode and saw that someone from 50.49.54.102 tried to register with my *. whois gives:- OrgName: Internet Assigned Numbers Authority OrgID: IANA Address: 4676 Admiralty Way, Suite 330 City: Marina del Rey StateProv: CA PostalCode: 90292-6695 Country: US NetRange: 50.0.0.0 - 50.255.255.255 CIDR: 50.0.0.0/8 NetName: RESERVED-50

I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or

Hi, Why does my asterisk keep forking instances at random times everyday? When I do ps aux, I got this: asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk -vvvg -c asterisk 23558 0.0 5.1 26040 12248 ? S 09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S 11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S

Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full -- Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold