similar to: Display name when dialing on Polycom

Polycom Provisioning Tool Updated. I made a bug fix that was reported, which was causing the directory creator to not work when there was an invalid character in the filename of the csv. I have also posted an FAQ: http://www.wintrisk.com/ppt.html#FAQ Download the new one, and tell me what you think! It's free, and mildly useful! http://www.wintrisk.com/ppt.html Yours, Michael Munger,

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:

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

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:

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

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 ;

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

This seems like something I should know... but.... I don't. How do you update zaptel / libpri on a Business Edition box running rPath? Tried running conary, but got 'Insufficient permission to access server conary.digium.com." Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com -------------- next part -------------- An HTML attachment was

I have a TDM800 that is installed and working. (TDM800 + 2 X QUAD FXO). Zttool says it is configured, ok, and there are no issues. Ztcfg -vvv shows that all the channels are configured. Zap show channels in the CLI show all 8 channels configured as they are supposed to be. When I plug in a pots line from the telco and make a call to that line, asterisk does not respond. (No Starting

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?

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,

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

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

I don't believe tinc will support this level of access control. As far as I can tell, it's all or nothing with tinc. How you configure firewalls on the other hand is up to you. On Tue, Oct 2, 2018 at 4:40 PM Michael Munger <mj at hph.io> wrote: > > Problem I want to solve: > > We have 3 sites: A, B, and C. > > Network admins should have access to all three. (this

Guys, this is a weird question but has anybody disabled the blind button that appears on polycoms or know if you can disable the use of blind transfers on polycoms to make any transfer attended? Thx!

You need more than an ATA. You need something with an FSO and FXO. I've used Linksys/SPA3102-3.3.6 and been happy with it. From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Sebastian Nielsen Sent: Thursday, March 21, 2019 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users at lists.digium.com> Subject: Re:

Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones....

Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR