similar to: Asterisk calls back after phone call

We have 2x BRI's connected to Asterisk which give us a total of 4 lines (using the bristuffed package). We would like to limit the number of incoming calls to 2 calls and if a 3rd call comes in, we would like this to go to another extension (voicemail or similar). Is this possible in Asterisk? Thanks -- James Doherty - Systems Engineer Zeald.com - Websites That Work! Web: www.zeald.com | Ph:

Hi all, In our asterisk box we have a Fritz card and an HFC card. So far I've had the Fritz card working well, but its time to connect the HFC card up to the ISDN line and get it working. I've followed the instructions on voip-info.org here: http://www.voip-info.org/wiki-Asterisk+zaphfc+install and here: http://www.voip-info.org/wiki-Asterisk+zaphfc+install But when it comes to

http://www.pcphoneline.com/skype "The VPT1000 is NOT a simple last generation USB phone audio device but is rather a next generation integrated gateway and USB phoneset with simultaneous dual mode Skype and SIP calling support. Skype is not forecast to have "SkypeIn" available until June 2005 but you can have the capability now via its built in SIP capabilities." Is this a

I have Asterisk RC2 setup here with a Fritz ISDN card on Debian Woody. I'm using chan_capi-0.3.5 and fcpci-suse8.2-03.11.02. Settings are pretty much the default, except in order to get the fcpci module to compile, I had to follow the instructions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install but the drivers would only compile if I left

I have a "Cologne Chip Designs GmbH ISDN network controller" in my asterisk box, and an ISDN line connected up to it. I downloaded the bri-stuff 0.1.0-RC4a package from junghanns.net and used that to compile asterisk using these instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc%20install When I try to call my asterisk box, my sip phone rings twice and then the

We have an office full of budgetones and find the ringing volume to be too loud especially when all the phones are ringing at once on an incoming call. I've tried setting the volume on each phone but the volume setting doesn't survive through reboots or power offs. Has anyone found a way to make the volume setting stick? Thanks -- James Doherty - Systems Engineer Zeald.com - Websites

In that example you could make outgoing calls only correct? (since incoming likely needs port forwards) I guess the questions becomes "how often are you going to do that to justify the extra $100 or so you going to pay for a wifi sip phone?" Paul Fielding (paul.fielding@shaw.ca) wrote: > > I think some people are missing the point. You can't throw your cordless >

I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11. It's relatively stable, and the last thing I want

I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32 C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25 Jan 18:25 Error Aborted 192.168.16.32 C:\Program Files\TFTP

Hello, I'm using Text::MultiMarkdown in my website, and I would like to know if there is any way to get syntax highlighting in code blocks. Thanks, -Mike Doherty

Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The

Hello, my name is Ikey Doherty. I am the founder of the SolusOS Linux Distribution. With SolusOS 2 we have decided to create our own base from scratch, using the PiSi package manager. After many repository revisions, it soon become apparant that GCC (4.7.2 in this instance) is dog-slow. Out of pure curiosity I attempted to rebuild packages using Clang. Clang had been pulled in as a dependency

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug

Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd

Can you recommend an R library that will help me create a diagram of a phylogenetic tree on which specific images are placed at appropriate nodes of the tree? For example, I have specific image files associated with each member of the phylogenetic tree, and I would like to automate the display of the image next to the correct node of the tree. I can provide an image file with an example.

Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometimes it doesn't, etc. I've come to the same conclusion as the wiki: it's probably better to avoid this

My problem is: No matter what machine I install and configure Asterisk on it I just can't get the music on-hold to work. Is anyone of you out there have such problem? If so what have you done to fix the problem? I've tried so far three other computers and none of them I was able to get music on-hold to work. CAN PLEASE ANYONE HELP. -------------- next part -------------- A non-text

On Sat, Mar 30, 2013 at 5:46 AM, Ikey Doherty <ikey at solusos.com> wrote: > So I tried building various other components with Clang. Notable (and well > known) failures include GLibc and the kernel. I find it more insulting that > the standard components put forth by GNU (such as glibc) rely on their own > non-standard extensions. Simply put: If you want to use one, you must

Hi, I would like to run R2.5.1 in Ubuntu feisty on an AMD64 chip. I am a total newbie to Linux. I have successfully compiled 2.5 however, I am not able to figure out how to upgrade the packages. I am not able to find an R-base-core_2.5.1-1_amd64 or all on the CRAN mirrors. I can find one on the Ubuntu feisty site, however when I try to install it I get a warning that Dependency is not stisfiable:

Hi all, I have been receiving the error "tree connect failed: ERRSRV - ERRinvnetname (Invalid network name in tree connect.)" for a while and I finally fixed it, but I have some questions about why it was happening. Firstly, it is Samba 2.0.7 running on Mandrake 8.0. My smb.conf file is at the end of this message. The machine's name is pinhead and in checking log.pinhead, I