similar to: install

Hi I was using the delivered Ubuntu 1.6.x packages but I wanted to look at gtalk integration so I downloaded, compiled and installed the source (after removing the Ubuntu packages) have installed the following: asterisk-1.8.0 dahdi-linux-complete-2.4.0+2.4.0 libpri-1.4.11.5 I copied my config back into place and most seems to work, but I cannot get my phone that is plugged into the Wildcard

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

i am running 1.4.37 and am hosted on Rackspace. I feel like a took a step back by using the Cloud server service since I am having a little trouble proving that my basic configuration is working. Nevertheless, I want to upgrade to 1.8. I use Centos 5.5 Anyone know of a good link that can help please? I searched Google and got confused by the options. Upgrade to 1.8. How please?

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

Hi, our Asterisk is connected to an E1 port. So we are using the DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for overlap digits for in-calls? I found the option "overlapdial=yes" but I did not try yet. Is that "my" option? Is there any option for setting an timeout? Thorsten

We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/4206395a/attachment.htm

I`m currently having aleatory call drops through a PRI and so I want to upgrade libpri. In case I have errors I will want to downgrade libpri, that`s the reason why I asked. Asterisk version: 1.6.2.13 Current Libpri: libpri-1.4.10.2 Libpri to be upgraded to: libpri-1.4.11.5 Any help will be thanked! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. I've been working with Asterisk for a few month now with great results. I'm very happy with this product. I hope I can be of help in the future of this project. Right now I'm having some problems installing it in a new server. I wont bore you with details because finally the solution to my problem was clean and elegant and it came from a guy in a Linux mailing list. Here is my

I don't think the original poster was asking about which OS is best. I think he was asking which PBX manager people are using. Ex, PBX in a Flash, Elastix, FreePBX, blah, blah, blah. Thanks; John From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Novack Sent: Wednesday, December 23, 2015 9:24 AM To: Asterisk Users

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms <http://voip.ms/> VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start

Hi, I’m currently using 32 CENTOS 5 but it’s now unsupported. I only have a 32 bit processor and CENTOS no longer supports 32 bit so I need to move on. I’ve installed the current version of 32 bit Fedora and I can’t get the latest Dahdi to build. Even tried downloading the early release DAHDI from github with no luck. Any recommendation for which 32 bit LINUX to use going forward? Or

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

What is the best asterisk platform to use? What are you guys using? I am looking for something to host either in our data center or at the customer prem where I have the control over the unit and not through a contractor. I dont mind paying a license fee for a front end interface but still would rather not have to pay. Thanks, --Eric -------------- next part -------------- An HTML attachment

Can someone advise on how to go about finding someone QUALIFIED to make changes to libpri? We have a pilot stuck on hold, due to old buggy PRI software on a meridian PBX. Upgrading the meridian software is not an option, so....we would like to have libpri changed to compensate for the bug. Is this a digium only type fix? I've called digium support but they only offer support for their

Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one

Hello thank you for your answer. If I read your (and others) reaction correctly I can conclude that this is an Asterisk problem and not a problem of MySQL or dialplan logic ? You should know that the MySQL database is heavily questioned : mysql> show status like '%onn%'; +--------------------------+--------+ | Variable_name            | Value  |

Frank Vanoni wrote: > On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > >> To that end, we’ve decided to discontinue the mailing lists effective >> February 1st, 2024. > That's a very sad news! :-( > Agree. Yet another giant step backward. Interesting that they will continue to send e-mails when postings to the (UGH) forum happen though. John Novack -- Dog

Hello, I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog POTS lines coming into my Asterisk server from the phone company. Internally, I have about 180 SIP clients defined in sip.conf. What appears to be happening is that if existing calls are consuming all 8 external lines and a new SIP client attempts to make a call, an existing call gets dropped. The asterisk

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: