similar to: Something wrong with the list?

Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm

Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk??

I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show

I just noticed the Aastra 57i do something that I haven't seen before. I called from one phone (phone 1) to the 57i. I answered it. Then, I pressed Transfer and dialed the extension for the third phone (in this case a Cisco 7960 in Sip). I did not answer the Cisco, but noticed the caller ID was showing the Aastra (as expected). I hung up the Aastra to complete the transfer and noticed the

I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. I've looked at the sample config files for 1.4 and nothing seems to jump out at me as to what

I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable,

Hello, >From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS.

Hey all, I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen: http://www.sineapps.com/news.php?rssid=722 The script, when run as the asterisk user, works properly and streams sound to stdin. But when Asterisk starts MoH it stops it immediately afterwards with no explanation. Has anyone

hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . try to call them !! i'll offer you some money . You can not Call them for some advices ... It's really a bad product don't waste your time to setup it. this enterprise must

Can a temporary message in Asterisk voicemail be de-activated so that the "regular" unavailable and busy messages are played. I have several users who are stuck with the temporary message. Thanks Mark

Dear friends, I followed the guide on http://techoctave.com/c7/posts/16-how-to-host-a-rails-app-with-phusion-passenger-for-nginx and successfully deployed two apps on the same web server, some months ago. Then when I upgraded to Rails 2.3.13.rc1 everything seems to be OK, but when I visit my app with the browser all that I get is the Welcome page from Nginx. Here is my nginx.conf:

Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP & NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked

Hi, due to some idiot not being able to configure his Exchange server correctly, this list has been swamped by loads of duplicate mails. There were still several hundreds of mails awaiting delivery to this list. I have removed all of those from the mail queue - so if you are missing a mail you have sent, please resubmit it, it probably was lost while I cleaned the queue as I had no way of

Hello, In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system. Is there a way to manipulate this message, as well? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Hi List, I wanted to know if anyone knew of a way with asterisk to "switch the voice" of a caller from male to female or vice versa. Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070426/2d483875/attachment-0001.htm

I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones

I will be out of the office starting 12/20/2005 and will not return until 12/27/2005. I''m out of the office and will return on Monday, the 22nd of March. If you have an emergency, please call 407.665.0311

Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Lacy's response in the thread 'Why does everyone seem to dislike *now?', has a small bit that caught my eye. Chan_Skinny made a lot of progress between 1.2 and 1.4, and even more in the later 1.4.X releases. I am curious as to which features/functions that chan_skinny might be lacking compared to chan_sccp. We (the community) now have a small, but active, group of volunteers