search for: sandman

Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you.

I'm attempting to set up a 2-way NT4-style trust between samba 3.0.1-2 (from source rpm on redhat 7.2) and Windows 2003 running active directory. The AD server (SANDMAN) trust of my Samba domain works fine, but when I try to use the Samba trust of the AD server I get the following error: [2004/02/04 15:52:34, 3] rpc_client/cli_pipe.c:rpc_api_pipe(462) Bind NACK received on pipe c009! [2004/02/04 15:52:34, 2] rpc_client/cli_pipe.c:cli_nt_establish_netlogon(1575)...

...ich is an NT server... Any help *much* appreciated... Thanks... Marc G. Fournier ICQ#7615664 IRC Nick: Scrappy Systems Administrator @ hub.org primary: scrappy@hub.org secondary: scrappy@{freebsd|postgresql}.org -------------- next part -------------- sandman:~$ smbclient \\\\relay\\dcurrie -U dcurrie Unable to open configuration file "/opt/samba/lib/smb.conf"! pm_process retuned false Can't load /opt/samba/lib/smb.conf - run testparm to debug it Added interface ip=131.162.129.111 bcast=131.162.135.255 nmask=255.255.248.0 Server time is Tu...

...ich is an NT server... Any help *much* appreciated... Thanks... Marc G. Fournier ICQ#7615664 IRC Nick: Scrappy Systems Administrator @ hub.org primary: scrappy@hub.org secondary: scrappy@{freebsd|postgresql}.org -------------- next part -------------- sandman:~$ smbclient \\\\relay\\dcurrie -U dcurrie Unable to open configuration file "/opt/samba/lib/smb.conf"! pm_process retuned false Can't load /opt/samba/lib/smb.conf - run testparm to debug it Added interface ip=131.162.129.111 bcast=131.162.135.255 nmask=255.255.248.0 Server time is Tu...

...risk_ip registration_port=5060 userlist_auth=globalpin config_auth=globalpin globalpin=1234 config_auth=disabled file_directory=/etc/asterisk/digium_phones [network] type=network alias=home cird=0.0.0.0/0 registration_address=asterisk_ip registration_port=5060 [Home_phone] type=phone full_name=Mr Sandman line=D70 ; this is the phone name I have in my sip.conf [D70] ; phone name I have in sip.conf type=line line_label=Line 01 exten=D70 ; phone name I have in sip.conf and extension number mailbox=D70 at default When I start asterisk, I can see : *CLI> == Digium Phone Module Started So that...

Does anybody know how to build a handset to sound card adapter ? Thanks! --------------------------------------- Norberto Harmath Centro de Comunicacion Cientifica Universidad de Buenos Aires

Hi guys, I've been experiencing a very strange issue with my Digium Card TDM400 as of this week. It has two FXS and two FXO. The FXO modules (both of them) never goes on-hook after hanging up in Asterisk. It had worked perfectly well for over four years. I put an ammeter in series with the line and the card, and immediately after plugging the connector to the card, I got 26mA in the circuit

Howdy All, I'm considering Asterisk / Digium as a replacement to my existing phone switch. I need to continue to be able to push analog lines between multiple buildings in a campus environment. The Digium Analog 410 Card manual states it's not recommended to go beyond 1500 feet distance for an FXS card, and no line should leave the building or be bundled. The 2400 Series Manual does

Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then

Can Asterisk be configured to hang up if another phone picks up? I'm a bit lost as far as terminology goes, but here's my setup. At home, I have asterisk answering calls from the pstn and sending them through to a sip extension or voicemail. All that is working fine. The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is

Hi Everyone, Looking to replace a condo intercom system. Apparently the current one taps into the lines and dials phone numbers but needs to be changed as it's faulty. I will probably still use the same analogue dialing and back it up with a VoIP line and use the current cabling that is in place. But as for as the door opening function goes, I am not sure how to interface and how open these

All, I recently got a request to deploy VOIP helplines with a bank's ATM site locations. The purpose of this solution was to be able to resolve cash dispensing errors as quickly as possible. My problem is that i do not have any idea about outdoor type VOIP equipments. Does anyone have ideas???

...11 11:43 AM > > Search the lists. Some hints: > Viking electronics makes a door box that connects to any analog line > (IIRC e-20). > They also make a DTMF keypad that integrates in series with any analog > line. They might also make a door box with a DTMF keypad on it. > Sandman makes a relay that will get energized when there is a ring on > the line which could be used to unlock the door. > > However, why would you use asterisk? Using asterisk for the sole > purpose of MDU entry system is like using windows for asterisk, it > works but why? > Go for...

Hi guys, I've had a sporadic problem recently with one of my users on our POTS line. About 1/3 of the time he dials a number (usually from a speeddial on his phone, I think), he'll get some phone company message (from the outside) about how the call could not be completed as dialed or something like that. However, the logs (and the console) always show the correct dialed digits.

Folks, I'm looking for a SIP or IAX phone for field techs to take with them when out on service calls. The regular desktop phones are just way too big. Is there anything like the size of a full-sized cell phone? Or smaller, not I doubt that... Tim --

Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to

...mple, and what the result would look like when played by ogg123. At first, I just meant for these tags to help out the Classical music side of things. But I've been assured that (other than Opus and Conductor) these tags are equally as useful in Pop music. Although with Metallicas "Enter Sandman" you might even need the CONDUCTOR tag, for the guy who conducted the cello players. -------------------------------------------------------- First, a few tags are now DEPRECATED: VERSION version information is typically included in the title, such as "Doctoring the Tard...