Re: [Asterisk-Users] Any IP phones with pro-audio connections?

I recently installed one of these plugged into an ATA with a dialplan that calls a predetermined number when the line is picked up. The sound guys have only to push one button and we get audio into the pbx. Works flawlessly. We did have to boost the input level more that I think we should have

[Asterisk-Users] Voicemail bomb

I submitted a bug to the tracker (bug)regarding the 256 character limit when copying a voicemail to a list of mail boxes. The bug was closed with this note: Fixed in 1.2 branch, merged to trunk. Could someone explain to me what that means... in English? I searched the release notes of the

Re: [Asterisk-Users] Re: Voicemail bomb

the fixed source code you can either: - wait for the next asterisk release or - download the current 1.2 branch from digium svn. Hope this helps... 2006/5/8, Garth Summey [EMAIL PROTECTED]: I submitted a bug to the tracker (bug)regarding the 256 character limit when copying a voicemail to a list

[Asterisk-Users] Voicemail indication for analog phones

I have 20 or so users with analog cordless phones connected via a 24port FXS box (vegastream). The vegastream supports voicemail indication via a studder tone, which is great but I have some users asking for a more positive (or proactive) indication of voicemail. I had a couple ideas; 1.If an

Re: [Asterisk-Users] Voipjet - No one is available to answer at this time

Don't think there is anything wrong with your setup. We get the same thing... Maybe they're down, but I would like a third opinion... G Michaël Gaudette wrote: Hi, I`ve just tried the Voipjet 0.25$ test account, following everything the web site told me to do (see below). When I dial a

Re: [Asterisk-Users] Testing AreskiCC

Not an answer to your questions, but just in case you don't know there is a lot of info on the wiki: http://www.voip-info.org/wiki/view/AreskiCC+CallingCard+Application We use Areskicc here, and it works great. However we do not use sip/iax friends, perhaps both of your problems lie there?

Re: [Asterisk-Users] Areski Calling Card GUI

If you haven't seen it already, this will be a lot of help to you. http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Application+The+idiots+guideV2 You should now be on step 12. :) G Omar McKenzie wrote: Hi I have gone thru the steps of installing AreskiCC, I

Re: [Asterisk-Users] wifi phones - desk

I've also only heard of the Clipcomm Along the same lines... Why doesn't anyone make a wireless ATA? Am I the only one with a need for such a thing? By the time I plug in a wireless bridge, an ata and a cordless phone, I need a five outlet powerstrip and shoebox to hide all the

Re: [Asterisk-Users] call to a particular 800 number never shows answered on Zap channel

This one drove me crazy for a while too. I found out that some companies don't exactly play fair and don't pass answer supervision on a call until you are actually speaking with a live person. The person I spoke to about this wasn't sure if that was even legal, but he said it happens quite a

Re: [Asterisk-Users] pins for users

We use the Areskicc calling card system as an authentication system. It does everything you are asking and can generate great reports and graphs. I like it very much. That being said, areskicc is tough to get going, but there is plenty of info here:

[Asterisk-Users] Voipjet experiment

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using

Re: [Asterisk-Users] Voipjet experiment

not their problem, but at the same time I would think they want to be as functional as possible. Thanks for the help everyone, G Garth Summey wrote: Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation

[Asterisk-Users] DND Indication

Hi, Has anyone come up with a clever way of indicating DND is activated? I've thought of stutter dial tone and using the mwi, but have no idea how to implement these. I'm using Budgetones. My concern is that users will activate the DND, then forget about it not realizing that they are not

[Asterisk-Users] app_intercept

Hi, Can anyone give me any information at all to get app_intercept working? I've found these pages, but there is just not enough for me to get it going. http://www.pbxfreeware.org/archives/2005/06/new_download_--.html and http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692

Re: [Asterisk-Users] [EMAIL PROTECTED] newbie extensions always busy

Let's start basic, we know that both PCs that are running the soft phones can see the aah server, but can both PCs see each other? Can they ping each other? (ie, they are not across a NAT router or something like that?) G Mark Anthony C. Delfin wrote: hi list, I'm running a newly

[Asterisk-Users] DND Indication

Has anyone come up with a clever way of indicating DND is activated? I've thought of stutter dial tone and using the mwi, but have no idea how to implement these. I'm using Budgetones. My concern is that users will activate the DND, then forget about it not realizing that they are not