similar to: * and Philips IS3090 PBX

Dear Mailing List, What does this error mean and how do I fix it? I am on a Centos 4.9 >From /var/log/maillog : Jan 14 11:54:51 hostname imap(username): file lib.c: line 37 (nearest_power): assertion failed: (num <= ((size_t)1 << (BITS_IN_SIZE_T-1))) Version information : root at hostname[/etc/rc.d/rc3.d]# dovecot --version ; dovecot -n ; cat /etc/*release* 0.99.11 Usage:

Hi Guys I have just come from a customer that is looking to install 13 Cisco CallManagers into all their branches, (i tried to convince them to go *). They are looking for a voicemail solution. Now as Kinesis and Unity are way too expensive (apparently cisco is launching a cheap voicemail system too) I was thinking of installing * as the voicemail solution. Lots of goggling i have found

Sorry if this show up twice, yesterdays post did not show up. We are having trouble setting up two ext's on one phone. We have it the point where the first two lines are ext 4000 then the third line is ext 4013. We can receive calls to both ext's but we can't make out going calls on ext 4013. The other thing that is strange is we need to have ext 4013 at the top of the config for ext

I will be out of the office starting 2004-12-24 and will not return until 2005-01-03. I am on vacation from 24/12/04 to 02/01/05 and will not have access to e-mails. I will respond to you on 03/01/05

I will be out of the office starting 2005-08-05 and will not return until 2005-08-22. I will respond to your message when I return.However you can contact me in kumar.pn@gmail.com ,Mobile (0)9848580960

Hi All, I have a Philips Webcam, model SPC110NC. I had to download a diver in order for the webcam to work with Skype from the philips support page (www.philips.com/support). Now when I start up TinnR to use with R, it opens the webcam. If I manually start up R, Tinn R will no longer synch with it. Also, when I install the webcam driver there doesn't appear to be anyway to modify the

Hi all, A few weeks ago, I sent a mail via the Philips website concerning their portable digital music players (Rush, Expanium, etc.) Below you will find my mail and the response from Philips. As you can read, they have no intention to support Ogg Vorbis in the near future. Perhaps if more of us ask for this kind of support, they will be interested in adding support. So, go to the philips

Hi there. R v1.3.0, Slackware linux 7.1 or Windows 98SE, PII 266, 64Mb RAM. I notice that under windows, R puts help info (ie. the response to '?<function>') into a new window, whereas under linux, it puts it into the main console window. I find the former much more helpful when checking a function's syntax. Is there a simple way of configuring R under linux to put its help

Does anyone know if version 3.1 is Call Manager or SIP? Thanks.

Hi all!, i search for some information about to setup my asterisk box with e-mail notification when a I call the voicemail application. Voicemail application works fine in the Dial Plan but nothing happens with email notification ...so what i need to know about this?...wiki pages did not help me ....thanks! G. ----- Original Message ----- From: <asterisk-users-request@lists.digium.com>

I am try to come up with a way to dial multiple ext and require one or more of the extension to require input before actually bridging the calls. Example: A call from PSTN into * a match is made in extensions.conf and * then dials a local(fxs) ext 1 and a Cell Phone If ext 1 picks up * bridges the call if Cell picks up require digit to in order to bridge the call if neither occurs and timeout is

hi, im using: --- passdb { driver = sql args = /etc/dovecot/dovecot-sql.conf.ext } userdb { driver = static args = uid=vmail gid=vmail home=/home/vmail/%d/ mail=mdbox:/home/vmail/%d/%n } -- this works fine on a test machine but i still see *2* instead of 1 connections to the pgsql db before a successful login is logged.. how can i avoid these extra connections?: Apr 17 11:34:49 merlin

I am trying to configure MeetMe so that external callers can enter the conference rooms after an IVR menu. I have created Conf rooms for all internal Ext's with a prefix of 8. When I call into the system from my vonage trunck the IVR picks up but will not let me dial a conf room. It tells me it is a invalid extension. Can anyone help with a sample conf on this? Thanks, RC

Steve Frampton [mailto:frampton@j-com.co.jp] of your address change Content-Length: 7142 samba@samba.org David Barroso <h4371719@alumnes.eup.udl.es> Jeremy Allison <jeremy@valinux.com> "Tulipant Gergely" <tulipant-gergely@dbrt.hu> Edwards Philip M Contr AFRL/SNRR <Philip.Edwards@wpafb.af.mil> Drenning Bruce <bdrenni@catholicrelief.org> Glenn

Hi.. We are facing a problem that is making the channel to be stuck. we are using asterisk 1.4.22.1 version, and we have a direct sip trunk...We have 2 queues and one has 2 agents and the other 5 agents, from last week the second queue's channel is getting stuck, it happened 3 times till now and the problem is calls come into the queue and just the calls will be in the queue and will not ring

Hello. My target architecture supports natively 16 bit integers (i16). Whenever I write in C programs using only short types, clang compiles the program to LLVM and converts the i16 data to i32 to perform arithmetic operations and then truncates the results to i16. Then, the InstructionCombining (INSTCOMBINE or IC) pass removes these conversions back and forth from i16, except for

Hi Guys I have been running a test server for a few days now with * 1.0.9 bristuff RC8n. I had a power failure and the test machine was not on the ups. When power was restored I found the following error: FATAL: Error inserting zaphfc (/lib/modules/2.6.11.4-20a-default/misc/zaphfc.ko): Unknown symbol in module, or unknown parameter (see dmesg) My dmesg output: zaphfc: unsupported