similar to: Off-Site Extensions That Would Show As In-Use?

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

I receive calls over a T1 with callerid and then *ani*dnis*. I am able to strip out the ani and the dnis in the dialplan but when I try to set the caller ID to be the ani, it looks ok but then if I do a NoOp callerid on the next line, I get unknown. Here is the section of my dialplan: exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN}) exten =>

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

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*

I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to be hands on for each new phone number deployed... so I would like to set up some administrative extensions that can record greetings... lets say: [admin] exten => 8(NXXNXXXXXX),1,Record($1|-greeting.gsm) [incoming] exten => _(NXXNXXXXXX),1,Playback($1|-greeting) exten => _(NXXNXXXXXX),2,Goto($1,1000) exten

Hi! I've got a number of extensions (about 50) on a working Asterisk setup. For each user, I have two extensions configured (for example 11021 for a Cisco 79XX phone and 11022 for X-Lite), and a ring group that ties the two extensions together (for example, 1102). Reason being that if the user is away from his/her desk or working offsite, they can answer the soft phone on the PC. From

Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. I'm sure someone has done this already. Anyone want

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Hi, I am interested to know the maximum number of resource pools that allow in Xen Cloud Platform. I cannot find any maximum number indicated in the documentation. Thank you, Wong Fui Fui _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

I am trying to use the value of an ID variable in an if statement and not getting the results I expected. # ID values for two school districts > with(rf, tapply(DistrictID, DistrictName, min) ) Aberdeen School Dist. # 58 Buhl Joint School District 59340 53409 This creates DNAME as I expected ...

xe vm-start vm=debian Your license has expired. Please contact your support representative. Network and Management Interface XCP 0.5.900-36486c I was seriously under the impression we fixed this, now all my images are offline is there any way to fix this without reinstalling everything ? _______________________________________________ Xen-users mailing list

Hola, What would a valid regexp in Asterisk be to identify a NANP number, i.e., NXXNXXXXXX? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:brent.torrenga at torrenga.com web:www.torrenga.com

I''ve been playing around with this, but have yet to be successful. I am trying to figure out how to SSH to my Dom0 IP and some static port and then redirect the "xm console <domU>" into the SSH session. Has anyone been successful in doing this? Really the only option I''ve seen is using "screen" on the /dev/pts/[1-x] console device, but that seems overly

Hello everyone, I have a simple histogram of gasoline prices going back a few years that I want to insert a line graph of consumer price index (cpi) over the histogram. I have looked through the "Lattice" book by Deepayan Sarkar but don't see anything there. How might this be done? An example would be wonderful. Current code snippet follows. For example additional field to add

Hi there I have the following setup Home - Main Tinc server with public IP running on PfSense work - tinc client running behind a CISCO ASA firewall with public IP running on Windows 10 offsite - tinc client running on tomato router behind a double NAT Home & offsite connect & i can see all PCs & devices & connect to them easily, on either side work to Home or offsite connects