similar to: Windows 2003 problem

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Thanks Curt, that's "too cool for school", any idea on when this is coming to the MS SBS platform? I use SBS for myself at home and would love that level of functionality included. Does Asterisk therefore handoff voicemail storage etc to Exchange for this level of integration? Cheers, Dean ________________________________ From: Curt Shaffer [mailto:cshaffer at

Found my answer for those who would like to know: Profile Rule: [--key $A]http://your.addre.ss/$B/$MAC.cfg GPP A: urtopsecretultrasecureaesencryptionkey GPP B: OddBallDirectory123098 Hope that helps someone! Curt -----Original Message----- From: Curt Shaffer [mailto:cshaffer@gmail.com] Sent: Wednesday, February 07, 2007 11:35 AM To: 'Asterisk Users Mailing List - Non-Commercial

I am having the same issue as below. Has this issue been solved or does anyone know an answer? This error recently began and we have multiple phones out of commission. PLEASE HELP!! http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -----Original Message-----

Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt

I wish to sync a bunch of flac files that reside in various subfolders to the root of a folder on a destination. An example of the directory structure on the source is: source> tree Music/ Music/ ??? R ? ??? Radiohead ? ? ??? OK Computer ? ? ??? 01 - Radiohead - Airbag.flac ? ? ??? 02 - Radiohead - Paranoid Android.flac ? ??? Red Hot Chilli Peppers ? ??? Greatest

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 10000-20000. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route

What is everyone out there doing in an all IP termination environment to change trunks when quality drops to a certain provider automatically? Thanks Curt

Sorry if this shows twice but it appears my first message was quarantined because of my digital signature. All, I have been tasked with setting up video conferencing utilizing asterisk. One of the requirements is a softset that has video capabilities. Eyebeam looks promising but I was just wondering if anyone out there knows of any freeware with comparable features of Eyebeam that they

I am having an issue with doing FTP auto provisioning of Polycom 501's when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router.

I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-?-vis. I know there is an option available for the Avaya systems but it?s a little out of the price range I?m looking for (~$200/channel). Has anyone out there found a stable way to do this?

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

I have decided to move on from Asterisk@Home and start compiling asterisk myself now. I got a dedicated box and put my X100P in it. I installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of O'Reilly's Asterisk the future of technology and begun. I downloaded the zaptel-1.2.4.tar.gz, libpri-1.0.9,

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using

Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/892a67b2/attachment.htm

I have read different posts over the months wondering who Vonage uses for their VoIP technologies. I stumbled across this article (although it's from 2002, I think) that suggests strongly that they use Cisco. There is no telling what they might use in conjunction with this but this should clear some of the conjecture.

I have been working with a couple companies who are interested in integrating Cisco VoIP (mostly call manager express) but utilizing Asterisk for AA, VM, failover trunks etc. I have found some materials and guidance out there but I think a list and/or wiki for general asterisk integration with other vendors would be great and feel that it is enough off topic that it deserves its own space. Just