similar to: conference and wifi phones

Hi list, I'm wondering if there's a way for multiple users to share the same voicemail box and have their BLF flashing when voicemail comes, i.e. in a home phone system where there's a general vm for everyone. I'm using couple Grandstream GXP2020. Any suggestions? Kelvin Chan | Positronics Ent. Product Development | | unit 272

Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St.

Hello, I'm trying to use the package ttda, wich is involved in text analysis, for my own data about answers in a company survey. I've installed it, as well as ispell, but when trying to use an example: > zz <- file("stupid.txt", "w") # build a data file > cat("{comment - stupid data file} \n" , file = zz) > cat("<uci=1>

I read the feature list of asterisk and I cannot see if it is possible to conference a call between extensions. Is it a supported feature of asterisk or is it done in the UA (ATA186 in my case) Here is what I try to do. phone-a -dial-> phone-b tap the cradle (flash on phone-a) phone-a -dial-> phone-c tap the cradle (flash on phone-a) Now I like all 3 phones in a conference call.

I was able to build Samba 3.0.x under Solaris 10 using the Sun Studio compiler (cc) and the Solaris ld (/usr/ccs/bin/ld) that it uses. Trying to build Samba 3.2.0, however, I immediately run into a problem where the build process assumes the linker understands "--version-script", which is specific to the GNU linker: Linking shared library bin/libtalloc.so /usr/ccs/bin/ld: illegal

I'm trying to build and install Samba 3.0.30 on a Solaris 9 SPARC machine. When I do a "make test", the RW1 test is failing. If I go back and configure and build 3.0.28 with the same settings, and do a make test, everything passes. Here's what I'm seeing with 3.0.30: ---8<--- Testing RW1 (0) TEST OUTPUT: host=127.0.0.2 share=tmp user=root myname=cannes Running RW1

With Samba 3.0.x, I would: 1. configure 2. make 3. make test 4. make install (if all went well) When I do a 'make test' with Samba 3.2.0, though, I eventually run into problems like this: RUNNING TESTS testparm_s3 Testing netbios name (0) TEST OUTPUT: Load smb config files from /home/david/build/samba-3.2.0/source/st/lib/smb.conf.tmp params.c:Parameter() - Ignoring badly formed line in

G'Day All, How do I fix this: I receive a call at the extension. Press the hold button. Music on hold starts. When I place the handset back on the cradle, the call gets hung up/disconnected. The Phone is A GrandStream Budge Tone 100. Thanks

Hi, Just got some CISCO 7960 phones. They seem to work great except if I make any SIP call using the speaker phone (leaving the hand set in the cradle)the call will disconnect in about 5 or so seconds. If I pick up the hand set and make a call, it's fine. Has anyone else run into this ? Any solution ? The phone is on SIP v6.1 - it did the same thing on 4.4 5.0 and 6.0. Thank you in

While pondering several issues, poor quality PSTN POTS lines, potential cost savings with multiple cell numbers, the FAX problems over TDM400p, etc, I wondered about: Cell phone "Base stations" to replace POTS lines. Devices to "cradle" cell phones and connect to TDM400p, for instance, to mimic PSTN. Are there such beasts, how do they play with asterisk? Will FAX work over

I have a Centos 3 server running Samba 3.0.28. It's a member of an AD domain on a Windows Server 2003 R2 Standard x64 SP2 box. From the W2K3 server I can see the samba share I created. Using the Security tab in the Windows Explorer file properties dialog I can add and remove users and change their permissions. However, in the Permissions tab of the Advanced Security Settings dialog,

> Well, I thought about it as well but up to now didn't try. Via which > way do you intend to have e.g. a line-in to the iPaq with reasonable > quality. Would be really interesting to figure out such things ... > and then use a WLAN-cradle to transport data to the internet *g* The main use would be covering events at sports fields, auditoriums, etc. that are away from the

Hi guys, I'm trying to move away from meetme to loose the dependency on dahdi. ConfBridge seems to be a good fit but I can't get it going. The document sounds like an easy to use app. Am I missing any bridge_ modules? Asterisk 1.6.2.0~rc2-0ubuntu1.2 -- Executing [521 at outbound:1] Answer("SIP/109-b877a8c8", "") in new stack -- Executing [521 at

I'm hoping somebody can point me to the right documentation for setting up the following scenario. Earlier this year I had Samba 3.0.28a working as a member server of a (Windows Server 2003) AD domain, using Solaris 10 and Heimdal Kerberos. I was able to log into the server using AD accounts, getent passwd worked, etc. I was using "secruity=ads" with these settings (among other):

Looks like it''s time for another release of Radiant: http://radiantcms.org/download/ Radiant 0.6.8 "Incise" adds a number of new features that enhance support for extensions and completes the project''s move to github. Of special note are: * Extension installer/uninstaller scripts (script/extension) and a global ''extension registry'' at

Hi. Does anyone know of any small SIP phones (and preferably have some experience of using them and happy to recommend them)? By 'small' I mean a single-piece phone, with dial buttons in the handset, so that it can be carried around easily in a laptop bag. Something like http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a standard analogue telephone).

Hi, I'm new to R (and math ;) Would somebody please be so kind as to direct me in plotting a 3D sphere? I tried something in the lines of: #### y <- x <- seq(-pi, pi, length=pi*10) f <- function(x,y) { z <- sqrt(pi - x^2 - y^2) #z[is.na(z)] <- 0 z } z <- outer(x, y, f) persp(x, y, z, theta = 120, phi = 30) #### I've also tried: .... make.surface.grid(...) ..

- This avoids problematic "reloc'ed while mapped" messages and some associated corruption as well. Signed-off-by: Maarten Maathuis <madman2003 at gmail.com> --- src/gallium/drivers/nouveau/nouveau_screen.c | 21 +++++++++++++++++++++ src/gallium/drivers/nouveau/nouveau_screen.h | 3 +++ src/gallium/drivers/nouveau/nouveau_stateobj.h | 13 +++++++++++++

Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use "canreinvite=yes". The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53