similar to: Upgrading AAH

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

All - I've been muddling around with this for a few days now.. and I'm trying to figure out why I am not receiving more than one phone call on each polycom 501 phone. I can make more than one phone call out, but not receive another one in, while on a call. Has anybody seen this behaivior before, or is there something simple in the config i'm missing, like.. maxcalls.. or

I've had Asterisk working on a test platform really well, but I've never found a decent web front end, that works in real-time. I've got a couple of incoming numbers that I'd like to have some IVR on (i.e. select this option etc), and then distribute the calls appropriately to various SIP agents, but also in some cases back out to a PSTN/Mobile number. I have this working

Hey all - I haven't run into this problem before, so I'm a little confused as to why I'm seeing it now. I've been testing with Polycom 501's in a ringall queue situation, and everything seemed to be fine. Now, if I make a call into the queue, and let it ring a few rings, then pick it up, the other phones setup in the ring-all queue, will keep ringing, even though another

Hi all, I have started my work of Samba authentaication usiing AD with Samba 3.0.7, openldap-2.3.9,kerberos 1.4.3 on Solaris 8. My first question is can i implement it on Solaris box? because where ever i see, i could see the document for Linux and Debian. Let me explain what i did. I compiled the Kerberos and LDAP package first. After that i compiled the samba package. Samba is compiled

On Sun, 12 Aug 2018, Blumenthal, Uri - 0553 - MITLL wrote: > Tone aside, let me second what Bob said. OpenSSH maintainers seem to > be able to find time for many updates and upgrades - but ECC support > over PKCS#11 appears to repulse them for more than two years (I don't > care to check for exactly how many more). There's no "repulsion" involved, just a lack of

Is it possible to use the outbound routing features of AAH0.9 but also allow a user to dial a prefix to force the use of a certain route? Thanks, Wiely -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050510/0c261294/attachment.htm

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,

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

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

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

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

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

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

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 =>

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

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

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

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