similar to: TDM400P - All extensions have same CallerID

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

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

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

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

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

I'm still a learning but I purchased a dph-140s to test with AAH 1.5. I think this is a rebadged ACT P104SLD which others seem to have working with *. It seems to be configured and registered similarly to the softphones I've been using just fine, but it does not receive or send audio (it will send audio to vm), or perform the loopback test. It seems to signal and receive calls fine. The

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; I've got an AAH installation where a customer wants to install an active Eicon DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel 2.4.21.37. Support for Eicon active cards is built-in. I've debugged and run the A@H install-Eicondiva script but when I try to run divactrl load -c 1 -f ETSI -Debug I get a response : A: can't get card type for DIVA adapter

Is it possible to access @resource variables inside a type? I would like to make some decisions on parameters based on other parameters that may have already been set. For example, --- newparam(:param1) do Puppet.debug "Found drivesperarray parameter" desc "parameter 1" validate do |value| if resource[:otherparam] then #dosomething else

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

Dear R users, I have noted a difference in the merge distances given by hclust using centroid method. For the following data: x<-c(1009.9,1012.5,1011.1,1011.8,1009.3,1010.6) and using Euclidean distance, hclust using centroid method gives the following results: > x.dist<-dist(x) > x.aah<-hclust(x.dist,method="centroid") > x.aah$merge [,1] [,2] [1,] -3 -6

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

Hy all need to preprocess 1) incoming call get caller id lookup some info in my db, 2) based on the result dispatch the call to the right operator step 1 is ok I developped a small .php script that connect manager and parse events, now I have to tell AAH do dispatch call to the right operator questions 1) is this the right practice ? 2) where to find a complete manager api reference, (to buy

Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to

Better read up on why a sip phone should register with asterisk. Do a 'sip show peers' and that will be the list of phones that can "receive" calls. ------------------- I've double checked this. Everything is logging in fine, because I can make calls, check my voicemail, everything except recieve calls on the SIP devices. David Phelan wrote:

I am running Asterisk at home on one of my systems here. It uses Centos 4.2. It does NOT have a GUI interface. Command line only at console or via SSH (yeah, I know that I was working on another server to get GUI tools working through SSH, but I am NOT going to muck with AAH build). yum update is potentially dangerous, when it replaces the kernel. You can loose all of your Zaptel driver

My question: How do I configure AAH via AMP to make a connection through our legacy PBX to the PSTN? Details: We're trying out Asterisk through Asterisk @ Home. Our legacy PBX has a modem type dial tone port that we hooked a Digium FXO to. Now I can dial from the XTEN client on my computer to any legacy PBX extension. If I connect a regular phone to the modem dial tone port, I can dial

OK. So check out the Wiki here.... http://www.voip-info.org/tiki-index.php?page=Asterisk The archive of this list can be search via google by entering... site:lists.digium.com <some parameter> www.digium.com has a link to all the materials for getting started in the Documentation section of the website. Those are really quite good so I would start there. Most were written prior to

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hello all, I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow something is not quite right with my vMail setup. I would have sworn this was all working, but maybe I was just dreaming. Anyway here is what is happening, say I am on extension 200 and I want to call to extension 201. If extension 201 is no connected, then it rolls right into vMail with the message the