similar to: AAH 0.9

Hi all, We use AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't

Howdy - I've been noticing a problem where I only receive a single call, before other calls go to voicemail. This only happens when the user is on the phone. I have the polycom 501's setup for 2 lines per key and 2 line keys for the first registration, which should allow for multiple calls. Anybody have any ideas what is going on? thanks Rolf Brusletto

Hi, Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone The dialplan was configured through AMP and has nothing fancy in it. As a first time user of not only Asterisk, but also a PBX, there are some operator teething problems. After much googling & searching of voip-info.org, I cannot find any answers to these

Hi everybody, I've proxy server IP, user ID and password. Now I need to connect to a remote Asterisk server as a SIP using my Asterisk @ Home box. That Asterisk server will make PSTN calls for me. I think I am making mistake while setting up the Trunk because when trying to make calls, it give all circuits are busy error. When I setup Sipura adapter, which is relatively easier to setup,

hey all, a followup from yesterday, not only are my incoming faxes blank, but they are also EXTREMELY small (like .4in wide), I've seen several people mention this problem in my searches, but no definative answer that works with the fax>email setup. Is there any resource that explains how AAH handles this or some tips for troubleshooting this issue? I've gotten several replies to my

any pointers to where this list is? I dont see it on the sourceforge pages. Hans Witvliet wrote: > aah-handbook (version 1.6) doesn't spill a single character about bri > and "tfot" doesn't spill much paper of the subject either ;-( > > Any suggestions/pointers > > Hans > You may want to try the AAH mailing list.

I am running AAH 2.8. I have an IVR for our main phone number that allows users to dial an extension directly. I would like to have a "this call may be recorded" announcement played before the call gets transferred. There is not a built-in option for this in the IVR web interface, but one way I can do this is to create ring groups for each user with announcements and modify the

Hello All, Has anyone successfully gotten the Click to Dial to work in SugarCRM in the latest AAH? I keep getting 'Invalid Channel' but I cannot figure out why. Thanks! Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050607/b18b3743/attachment.htm

Hi, Can anyone please point me in a direction as to how to set up these 2 pci cards with AAH 1.1? I have (am still) googling left, right & center - but haven't found a definitive guide yet. The centos based setup lacks any of the tools I know (insmod, modprobe ....) so it is time consuming just to even dig around the AAH box. There are no zaptel.conf files ....and on it goes. A

Hello I have couple issues with AAH. 1.5 1. Flash panel doesn't show proper status. Sometime accessing with IP seems to work and it shows current line status etc. Sometimes accessing with hostname of the asterisk server seems to show lines, but it doesn't show off hook etc when we pickup a extension and talk. In /var/www/html/panel/op_server.cfg I have tried setting manager_host to all

Hello, I use Asterisk@home, i see i've the version 1.1, the version 1.5 is released. Is it possible to upgrade my "old" version to the new one without reinstalling all the server? Is it possible to just make a update? Thx Jme

Using some scripts that have been posted, we have been able to get paging to phones working quite nicely. However, with a few Asterisk@Home 2.5 installs, (Aserisk 1.2.4) the phones ring but never pick up. Any ideas on why or how to tweak the scripts to get the phone paging working again? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider

All - I've a new system, that since it's been in production, has seen a few issues, that look like they should be fixed by upgrading asterisk @ home to the latest version. I was curious if anybody out there can tell me their experiences with this, and what to expect. Thanks, Rolf Brusletto

Ah - 'yum install mysql-server' It's getting late! Thanks anyway

Does anyone know if FreePBX can be installed on a Linux box that was built using Asterisk@Home. I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only

Hi, using asterisk@home, with quadBri from junghanns.net I am facing a strange problem: I have set incoming routes for some extension / DID: [ext-did] include => ext-did-custom exten => 23,1,SetVar(FROM_DID=23) exten => 23,2,Goto(ext-local,23,1) exten => 57,1,SetVar(FROM_DID=57) exten => 57,2,Goto(ext-local,57,1) exten => 66,1,SetVar(FROM_DID=66) exten =>

Hello all, I am having trouble getting my IAX based Voip provider setup. Any pointers are welcome. So here is the deal. I am registered up and I can make outgoing calls but incoming calls fail. Configs all look good I thought. My PBX is behind our firewall with a direct NAT of one to one for an external IP. IAX port is forwarded UDP and TCP to the internal IP. * shows good registration and

Hi folks. Thanks for having patience with a newbie... I am going through the setup exercise at http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm I am trying to do it all on BroadVoice since this is just experimental, and the 100 minutes of outgoing from them should be enough for now. So I set up BroadVoice for both inbound and outbound, using a combination of the instructions referenced