search for: creasy

Displaying 20 results from an estimated 58 matches for "creasy".

2006 Apr 19
2
PRI caller ID
...00 a1 17 02 01 01 02 01 00 80 0f 43 65 6c 6c 20 50 68 6f 6e 65 20 20 20 4b 59] < Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'Cell', 0x20, 'Phone', 0x20, 0x20, 0x20, 'KY' ] -Jonathan Jonathan Creasy Network Engineer BluegrassNet Development www.bgnd.com www.bluegrass.net o. 502-589-4638 c. 502-889-5567 h. 502-541-0566 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060419/2df6a818/attachment...
2005 Sep 12
5
What have I misconfigured?
I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP
2005 Aug 11
2
Is it mandatory to give power supplytoTDM400Pcard
...@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith Sent: Thursday, August 11, 2005 9:44 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Is it mandatory to give power supplytoTDM400Pcard On Thursday 11 August 2005 09:30, Jonathan k. Creasy wrote: > Ok, I just unplugged my power connector to a card with 4 FXO modules and > they no longer work. You're *sure* you've got FXO modules and not FXS ones? FXO plug into regular phone lines, FXS plug into telephones... Unless Digium changed something and now powers both off o...
2005 Aug 31
5
Asterisk for Voicemail Server
How does one go about connecting Asterisk to a Meridian PBX to handle voicemail? I have a customer who is out of capacity on their voicemail system (which connects to their meridian via several FXS cards) and I would like to see if I could use Asterisk to handle their voicemail. -Jonathan
2005 Aug 11
2
Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone
...erisk-users-bounces@lists.digium.com] On Behalf Of Andrew M Stemen Sent: Thursday, August 11, 2005 5:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone Jonathan k. Creasy wrote: > Yeah....I think that every install I have done the first thing that > happens is "why is there a delay before the call connects?" and the > answer is "you have to hit dial or wait 10 seconds". What all phones does that apply to? I'm fairly certain it appl...
2005 Sep 13
1
populating asterisk realtime tables from configfiles
...ql); if(DB::isError($result)) echo $result->toString(); } } } ?> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan k. Creasy Sent: Tuesday, September 13, 2005 8:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] populating asterisk realtime tables from configfiles Does any one have a script (I prefer PHP) that reads the config files, parses them and inserts data into the asterisk...
2005 Aug 11
4
Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone
...erisk-users-bounces@lists.digium.com]On Behalf Of Andrew M Stemen Sent: Thursday, August 11, 2005 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone Jonathan k. Creasy wrote: > Yeah....I think that every install I have done the first thing that > happens is "why is there a delay before the call connects?" and the > answer is "you have to hit dial or wait 10 seconds". What all phones does that apply to? I'm fairly certain it appl...
2005 Sep 06
2
Polycom ip301 hangs at Running "sip.ld"
My polycom phone is now hanging at Running "sip.ld". I modified it's config via the web interface to register with my asterisk box. I have tried to restore the default settings wth 468* and it doesn't seem to work. Any ideas? -jonathan
2005 Sep 09
2
call volume
Just a poll because I am curious, what kind of call volume do you see? Calls/Day -Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050909/9b8adf09/attachment.htm
2006 Feb 01
2
Dundi key Problem
I am getting the following message when trying to lookup up a number via Dundi: Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key 'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for '00:a0:c9:55:91:89'! I have created keys on each box with "astgenkey -n office.pbx.bluegrass.net" using the host name for each box of course. I
2005 Jul 13
5
CONSOLE/dsp
I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for
2005 Sep 17
22
AstriCon 2006 Location
The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com
2005 Aug 17
4
How many TDM22P Card can be used on the same PC ?
Just Google the archive on 'IRQ issues'. You can pretty much bet that 6 TDM cards on 6 PCI slots would suck hugely. Unless echo is your goal, you are not going to be pleased. If you have to use 24 existing POTS lines, look into a channel bank and interface it to a T1 card. If you are planning new, just get a PRI T1 and be done with it. Cheers, W -----Original Message----- From:
2005 Oct 15
1
No Audio from Console but mpg123 from shell worksfine.
Anyone have anything on this? (I'm sure someone will complain about me bringing it up again, chill out...) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan k. Creasy Sent: Friday, October 14, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No Audio from Console but mpg123 from shell worksfine. I get audio from mpg123 at the command line but when I load up asterisk and try to get audio from the console it look...
2005 Sep 09
9
adding DNIS digits
Situation: 8 POTS lines, 3 companies, 1 system. Channel banking the POTS lines onto a T1 thru an ADIT 600. The only way our carrier will provide DNIS is thru Analog DID #'s. Anyone know of a piece of hardware that can add DNIS digits to a particular line? -Darren
2022 Sep 29
1
[Bug 3477] New: Support environment variable or %u token for User
...ser Product: Portable OpenSSH Version: v9.0p1 Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: ssh Assignee: unassigned-bugs at mindrot.org Reporter: ben at bencreasy.com While I realize most of the ~/.ssh/config files are bespoke, I'm trying to write a shared one for my team. Right now I'm just leaving the User field out which means that instead of people typing ssh <server> they type ssh <user>@<server> - I'd like to save them s...
2005 Aug 11
1
Firewall will definately increasejittersinyourvoice conversation
...risk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Esben Stien Sent: Thursday, August 11, 2005 8:11 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Firewall will definately increasejittersinyourvoice conversation "Jonathan k. Creasy" <jonathan@bluegrass.net> writes: > even when I have taken all other security measures I also lock down > a box with a firewall You still should follow the lists. As long as your computer is processing data received from the net, it may be vulnerable;) Generic attacks against th...
2005 Sep 30
1
(no subject)
My Polycom IP301 hangs on "Processing Cfg..." Here is the boot log: 0930155446|so |4|00|---------- Initial log entry ---------- 0930155446|so |4|00|+++ Note that bootrom log times are in GMT +++ 0930155446|wdog |4|00|Initial log entry 0930155446|cfg |4|00|Initial log entry 0930155446|copy |4|00|Initial log entry 0930155446|cdp |4|00|Initial log entry 0930155446|cdp
2005 Oct 16
1
No Audio from Console but mpg123 from shellworksfine.
-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, October 16, 2005 2:59 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from shellworksfine. >One possibility is that the volume is set to 0. aumix can be handy here. Does
2005 Oct 17
0
No Audio from Console but mpg123fromshellworksfine.
...s-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of brett@websmyths.com Sent: Sunday, October 16, 2005 9:00 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] No Audio from Console but mpg123fromshellworksfine. On 10/16/2005, "Jonathan k. Creasy" <jonathan@bluegrass.net> wrote: >-----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tzafrir > Cohen > Sent: Sunday, October 16, 2005 2:59 AM > To: asterisk-users@lists.digium.com &g...