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Hi, I've been using TrippLite's PowerAlert software with my RM2200XL2U rack system and my APC BackUPS desktop systems. Unfortunately it doesn't work with openSUSE 10.2 (the software has problems with the serial port and UPS interfaces on openSUSE 10.2), so I've been looking at other options. I see that openSUSE 10.2 includes NUT, and it appears to work with my TrippLite UPS via

I keep getting this kind of stuff in /var/log/messages Jan 18 17:45:03 bulldog tripplite_usb[9979]: libusb_get_interrupt() returned -110 instead of 8 Should I be worried about it? upsc seems to work fine... I tried building the 2.0.5 but none of the included spec files seemed to work with opensuse 10.2 and adapting their 2.0.4 spec file didn't seem to work out well either. Building from

I keep getting this kind of stuff in /var/log/messages Jan 18 17:45:03 bulldog tripplite_usb[9979]: libusb_get_interrupt() returned -110 instead of 8 Should I be worried about it? upsc seems to work fine... I tried building the 2.0.5 but none of the included spec files seemed to work with opensuse 10.2 and adapting their 2.0.4 spec file didn't seem to work out well either. Building from

I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give.

I search Google to find how to get the message light to flash on my Cisco 7960 running (Application Load ID POS3-06-3-00) (Boot Load ID PC03M030) (DSP Load ID PS03AT38) All I see is about the sip.conf file witch mine has the mailbox=XXXX but still no light. Also the messages button does not work. Any ideas?

I have two domains, one called CORP (which is 2003 AD) and the other is called LABS (which is running Samba 3.5.4-5.1.2). I have established bi-directional trust between them and most of the basic tests show apparent success. However there are a couple of issues left that I am trying to chase down. First issue is that I would like to filter out the local (LABS) users and groups in winbind if

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

Is user authentication ever needed for just querying variables? My cacti script seems to be cooked, and I'm just wondering if I need to mess with username/password stuff. Not hard but if it's not needed then why bother... Thanks -- Eric A. Hall http://www.ehsco.com/ Internet Core Protocols http://www.oreilly.com/catalog/coreprot/

...60 500 CARDS: I have created some custom numbered cards. So, every call started with the pattern "63" goes to trunk or route Gamma with a price of 0.60$ and a connect fee of 5$ I hope this helps you. Regards... Nahuel Ramos. On Mon, 29 Nov 2004 23:30:35 -0500, Eric Hall <ehall@amaxx.com> wrote: > > Hello Sir > > Could you send me a copy of your route Pattern? I'm having trouble > > > > Thank You

Group I'm using realtime for voicemail the it works great.. The only problem I have is I'm not able to use directory or vmail.cgi Does anyone have a solution for this problem? Asterisk CVS-HEAD-01/24/05-07:36:37 RedHat 9.0 Any help would be great!!!! Thanks

...Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you already get the info. If you get caller ID number, then your telco is not sending the name. On 5/1/06, Hall, Eric M. <ehall@amaxx.com> wrote: > > Do you wait before or after the answer? Do you even need the answer? > > > > > -----Original Message----- > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com<Alex.Lopez@OpSys.com> > ] > Sent: Mon May 01 14:26:49 2006 > To:...

I'm looking to set up asterisk to call customer 3 days before the app and remind them we will be out to see them. I'm looking for any ideas on good ways to do this. Also I think it would be best to do some type of text to speech however I do not like the sound of the free one . Any ideas? Thanks!!! Eric Hall -------------- next part -------------- An HTML attachment was

D Not sure why this works exten => _3665[0-9],1,goto(test|${EXTEN}|1) but this does not. exten => _366[50-59],1,goto(test|${EXTEN}|1) I would like to route 36650 - 36700 to a Context 'test' however I'm only able to get 10 to work at a time. Any ideas? Any help would be great! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid

On 1/2/07, Eric A. Hall <ehall@ehsco.com> wrote: > I can hack up a Cacti PHP script to monitor UPS state via the upsc client > in the meantime. I'll have to manually enumerate the UPS names associated > with a host target (is there a way to probe for them with upsc now?), but > otherwise it ought to be pretty...

Hello group I just installed ASTCC and it was VERY easy to get running. I have a question about Pattern Via the web page I click the Routes link and everything makes sense to me but the pattern part. I tried _NXXNXXXXXX with the idea that everything would match this. Well it doesn't work... Does anyone have a good how-to? Thanks for all your help!!

Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI> -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1

Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G

...used the same images with MagicISO multi-boot (Windows utility) and it loads fine there (although using isolinux to load the Linux System Recovery tool craps there, alas). ----------------------------------------------------------------- Here's the command and output for building the ISO: [ ehall$ ] mkisofs -R -J -N -b isolinux-debug.bin -c boot.cat -o /tmp/rescuecd.iso -no-emul-boot -boot-load-size 4 -boot-info-table . Warning: creating filesystem that does not conform to ISO-9660. INFO: UTF-8 character encoding detected by locale settings. Assuming UTF-8 encoded filenames on...

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has