similar to: AAH Mailing list

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

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

Hey all, been noticing some oddness on a new AAH install... occasionally an incoming zap line with automatically connect with an outgoing extension, even though the incoming line hasn't specified what extension it's aiming for (i.e. haven't tapped in the ext # yet)... so someone's trying to call out from inside the office & are automatically connected with an incoming line.

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

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 all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

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

Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance T1), our current pbx has the signaling set to E&M, I can set em in zapata.conf, but I'm trying to track down the proper entries for the zaptel.conf file. The digium docs only show a PRI example. Our current system has these settings: Signalling: E&M Framing mode: ESF Line Coding: B8SZ here's my

Is there an easy way to disable flash transfers? I'd prefer the users hit # to transfer, since some users are hanging up a call, then dialing another one without giving the handset enough time to actually hangup the call, so it appears that they are transfering the 'ended' call to the new number that they are calling.. I'd like to keep flash functionality for call waiting, but

Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting assembly? I can get one to work, but when I install two, one always fails. I've tried all my cards solo in the enclosure, on each side, and they all work properly when only 1 is installed, however, when I install two, one of them will come up, but the other always fails. Anyone know what might be causing this?

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

30 says it's view only in the docs & I can't seem to change it, any other options? > Option 30 allows to set Module Shelf Address/ID.

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,

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

Hey all again, I'm wrestling with echo problems on our sip extensions. I've set these items in zapata.conf but tweaking these values doesn't seem to make much difference echocancel=yes echocancelwhenbridged=yes echotraining=2500 rxgain=8.0 txgain=1.0 are there other settings that can help me tame this beast? Been searching but not turning up anything that'll work here. Thanks

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

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

Hi all, been searching & not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. Any clue on how to make it ring twice before getting picked up? I tried immedate=no and

Hey all, I've been seeing this repeatedly and am wondering if anyone has a clue what's causing it.. at least once a day I see two zap fxo channels being bridged, and hanging..now, these two channels should never bridge, but they keep doing it.. any leads on where to look for what's causing it? here's what I see when I 'show channels' Zap/37-1 s@from-pstn:1

hey all, trying to get a zap extension to work & I can dial out normally with it, but if I try to access any of the features (i.e. *97 for voicemail) the zap channel doesn't hear it, and all i get is dialtone. Is there a dialplan setting or something to make the zap channels recognize keys like * or # ? Thanks in advance