similar to: [OT] Oldest Telephone

I'm new to this list. Reading the asterisk handbook pdf (good work) but but still have some questions. Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri. We have a dedicated server which is connected to our telephone company. It makes us able to call ordinary phones via VOIP using Ericsson DRG22. Would like to make people able to call me - and get a message "dial 1

Hi Elliot, I couldn't find anything related to smbpasswd expiry. Since u have "unix password sync = true", just a wild guess, if u could turn off password ageing in unix passwd file (man passwd for more detail) and see if problem persists. Rgds Gary Elliot wrote: > Hi guys... I setup samba 2.2.5 as a PDC ... I have w2k clients. It seems > that now I am prompted to change

Hello list, we are experimenting with using Dovecot as part of a simplified mail stack with SQL for a internal messaging / notification system for our customers. Sometimes we need to send a message to thousands of customers at once and we do it by having aliases resolve to 10k and 20k+ mail accounts. When we send such a message the server load (obviously) is very high for a few minutes (in case

Hello, I have a problem with my samba installation I can not get my head around, maybe some of you have a good idea about what is going on. I have a file share called "adfs02" and an AD DC called "addc02" in the same site. The error occurs only with this one user, and it worked til the last password change of that user two days ago. Here are the outputs of my test case (both

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Yes you are right, i realized that just after i sent the email, that it was a bit in haste. anyway, if there is anyway how to optimize dovecot for effective delivery of 20k+ emails within a few minutes, id be glad to know. .. and i will have to ask the same q at postfix mailinglists gnd/ On 7/14/20 8:12 PM, John Stoffel wrote: > > gnd> we are experimenting with using Dovecot as part

Hi everybody, I've got an Unitek iZi UPS 525 with a serial interface. The supplied cable looks fairly simple: RX->RX , TX->TX and GND->GND. Using this cable and Minicom with 2400 8N1 I'm able to issue commands to the UPS using the "Megatec Protocol" http://www.networkupstools.org/protocols/megatec.html so I know the UPS is fairly "intelligent" :-) and the

I've been playing with the outgoing call spooling feature a bit lately and it all works as it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm

Hi people, I was wondering if the IE Tab Firefox extension would work on wine. Has anybody try it yet? If someone did, please let me know how it worked. Here is a link to that Firefox extension for those who do not know it yet: https://addons.mozilla.org/addon.php?id=1419 Thanks a lot -- .''`. John Doe : :' : LUGS: http://www.cafelug.org.ar/ http://www.lugsam.org.ar/ `.

Hi, I'm trying to setup Asterisk on a Linux (redhat 8) machine. There are no analogue phones to be used & use is purely for internet traffic (SIPs). I've followed the setup guide from : http://www.automated.it/guidetoasterisk.htm But cannot get asterisk to run. If I type asterisk at the command prompt I get invalid instruction If I type ./asterisk start I get [FAILED] Can

I have my new phone mostly working. I do have a couple of residuals that I cannot find mentioned in the list archives: 1. Is it possible to set the volume in these things? I hope I didn't miss it, but I've looked in the doc, the FAQ, and the asterisk archives and don't find anything. The displays in the pictures all have more bars on them than my phone does, and I need a bit

On Thu, 2005-07-28 at 08:44 -0500, Bryan J. Smith wrote: > On Thu, 2005-07-28 at 08:44 -0400, Peter Arremann wrote: > > *sighs* BS level exceeded... > > DVD-RAM is the most reliable, long-term optical archiving format. Well other than the name, it does not have anything in common with DVD because IIRC it's a Magneto Optical drive ... totally different Tech. Paul

It there a way to get the genericups driver to set a output on the serial port? On the APC BackupPro the will pull the BattLow line low with a pull down resistor it's up to you to from the PC side to pull the line high with a pullup resistor from one of the RS232 output pins. Then monitor the line with another pin. Regards Rabie -------------- next part -------------- An HTML attachment was

Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has

Hi, I have resources available to host a portal specifically for the Asterisk system, to help correlate documentation, FAQ's and How To I am new to Asterisk, and my hardest work is in locating information on using or configuring the software. Would Mark, John or any of you feel this would be of benefit to host? I am offering a PHP-NUKE portal for this, (Unless you know of a better

I know in the x100p hardware the driver can sense whether there is battery voltage on the line or not. Is this possible in the zap drivers using t1/e1 interfaces ? I believe the signalling bits are only ring and hook, not loop presence, so not sure if it is possible. I had a situation last night where a main cable was cut in the area and lines were down overnight. My burglar alarm knew right

We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS

ACES - Asterisk Communications Endpoint System {the following could be used by any IP-PBX but the name pays homage to Mark Spencer and friends who cannot be lauded enough for their fine work} As you read this it will be obvious I am not a professional engineer but I do have enough knowledge to be fairly certain what I'm proposing is feasible from not only an engineering, but production