similar to: New Polycom Provisioning Tool Released with BugFix

I have had so many requests for it, I have released the source. http://www.wintrisk.com/ppt.html Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> -------------- next part -------------- An HTML attachment was scrubbed... URL:

Will this work to match any number from the 770,404, or 678 area codes? _[404|770|678]NXXXXXX If this won't work, is there a pattern that will do this? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> .

Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> . -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to use PHP to reload the extensions in an Asterisk installation. I keep getting this error: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) when I run the script by visiting the URL; however, if I run the script from the command line, it runs just fine (works perfect, actually). I think it is permissions related. Does anyone have any ideas? <php

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

We have an asterisk system and Polycom phones that were provisioned by someone else. They want to get call waiting to work, but for the life of me, I cannot figure out why the Polycom is returning a SIP 486 Busy Here when you call and the person is already on the phone. I have the feeling there is a configuration in sip.cfg or mac.cfg that I am overlooking. Any thoughts? Calls per line key

What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of MOSBAH ABDELKADER Sent: Saturday, August 04, 2007 3:16 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Connecting

Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says "Request sent." The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but

Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal ____________________________________________________________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today!

I have a customer who wants the Polycoms to display the CallerID name of the person they called on the phone they are calling from. The receiving phone gets CID just fine, but the calling phone doesn't display a name. For instance, if you dialed extension 3000, the Polycom Displays "3000(3000)" Instead of "John Doe (3000)". How do I set that up? Yours, Michael

Just got one of these. Horrible to program. Trying to key in the FTP server. Won't even remember the info after rebooting. Anybody know the proper way to beat on this stupid beast so it will work?

Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're

I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

I am trying to get a zaptel install converted to dahdi. I can get dahdi installed, and the pseudo device even shows up; however, dahdi show channels shows me nothing. There is a TE122 and a TDM800 in there, and neither show up. dahdi show status shows both cards, and dahdi tools show that the cards are there, working, and have no alarms. What am I missing? Michael Munger, dCAP, MCPS, MCNPS,

Is there any way to obtain the public key from the private key? I know it's in host file however for the purpose of this message I need to be able to generate it on the command line. I tried openssl ec < ed25519_key.priv I get an error read EC key unable to load Key 140092556813984:error:0906D06C:PEM routines:PEM_read_bio:no start line:pem_lib.c:703:Expecting: ANY PRIVATE KEY

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

Hello all, my apologies if I have failed to RTFM, but the PDF manual and google were no help. I am trying to get a simple two node tunnel setup. I have verified that the up scripts bring up the interfaces as they are written; however my gut feeling is that there is something wrong there, but I am not sure exactly what. Misc items: * I have a Port line in horcrux because I ran INIT as a user,

Does anyone have an (overhead) paging system that they like that works with SIP? We've got a client with an old paging system that (supposedly) just takes an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn't auto-answer the call, so paging never happens. [cid:image001.png at 01D4DFF6.9C1F1AA0] Michael J. Munger, dCAP, MCPS, MCNPS, MBSS Microsoft Certified

*Problem I want to solve:* We have 3 sites: A, B, and C. Network admins should have access to all three. (this works as-is). Desktop support should only have access to their site. (Tech A to site A, Tech B to site B, Tech C to site C). *How I think I can do it:* Working with keys? Admin's public key will be on all the client machines, and thus, the client machines will always