similar to: FW: Why does One-Touch record mute/disconnect call if not dialed quick enough?

Hi, Any suggestions to below problem? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jamie Heckford Sent: 17 October 2006 21:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect callif not dialed quick enough? Hi List, Have an odd problem

Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant

Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left

Dear developers: I have observed a change in the behaviour of S3 method despatch (as I guess related to namespaces) between 2.6.2 and yesterday's 2.7.0 RC and would be grateful if you could comment on that: the 'Axis' function in the 'graphics' namespace calls UseMethod("Axis",x), internally to despatch on the S3 Axis method depending on the type of the argument.

Hi, We have quite happily been running dovecot with no issues at all for approx. two years now and have had no problem whatsoever. Great work :) BUT (there is always a but!).... yesterday one of our users started complaining that they were having issues with their mail. I checked and although they can still view their IMAP mailbox ok it constantly throws up error messages such as: "The

Hi, Damn I hate upgrades. After updating at the weekend to FreeBSD 5.4-R in a fresh install our samba server just doesn't want to play ball. I have managed to get Samba3 working ok apart from when it comes to using Excel documents. If a user has an excel document open and another user tries to open it, Excel 2003 will fail with 'Unable to open the file, it may be read-only or

Hi, Is it possible to add users/change passwords for users in samba without it being dependant on anything in the UNIX password file? For example I want to add a user to my samba password file without them existing in the UNIX password file. If anyone has any RTFM info they can kick me in it would be most appreciated :) Thanks -- Jamie Heckford Network Manager Trident Microsystems Ltd Tel:

Following my previous post on S3 method despatch, I put debug messages in the code of Axis, Axis.default and plot.default in graphics/R/axis.R and graphics/R/plot.R to print the class of x, at and y on plot. After recompiling R, what I see is that x *lost* its class attribute (at least for classes not known to 'graphics') in Axis, called directly from plot.default and this could be the

Hi all, I'm seeking feedback (good, bad or indifferent) in regards to developing (further) a new class system for R, that uses encapsulated class definitions (i.e. the method definitions are literally inside the body of the class definition). A working (however very rough and poorly tested) system is available in my R package "oosp" with documentation in the vignette

If I have a class Foo, then i can write an S3 method for sum for it: >setClass("Foo",representation(a="integer"));aFoo=new("Foo",a=c(1:3,NA)) >sum.Foo <- function(x,na.rm){print(x);print(na.rm);sum(x at a,na.rm=na.rm)} >sum(aFoo) But how do I write an S4 method for this? All my attempts to do so have foundered. For example

Full_Name: Mike Kay Version: R-patched OS: Linux Submission from: (NULL) (137.75.70.37) Hi, The following patch cleans up some grammar in the docs for UseMethod (library/base/help/UseMethod) -mike --- UseMethod 2005-09-28 20:06:39.000000000 +0000 +++ /tmp/UseMethod 2005-10-31 21:21:05.534708720 +0000 @@ -5,7 +5,7 @@ Description: R possesses a simple generic function mechanism

Hi, I'm having problems with an occasional disconnect from phone calls while my phone is on mute. This is a problem with long conference calls, for example. I've a GrandStream GXP-2000 and Asterisk 1.2.1. Anyone have experience with similar issues? Best, WILL

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Hi All, We are currently evaluating the confbridge application while we prepare to upgrade our environment to asterisk v1.6.2.x. We have run in to two issues using it to kick/mute participants in a bridge and would like to ask for the experience of others running the application for any work-arounds. Firstly for kicking participants, would it be possible to use the softhangup application

Hi. I am using latest asterisk 1.2 and it would be nice in a meetme conference to be able to mute all but a particular user and then unmute all those users again with one command. Am I missing something or is this not available? Maybe I could write something, but I wanted to check first. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?

Hello ppl, Is there anyway to control a call mid-way in terms of sending a re-INVITE with say sendonly, etc. to mute one call leg of a bridged call ?? Looked around, so far, doesnt seem to be possible. If it's not, I think it's quite an important feature (re-INVITES mid-session) for a B2BUA. cheers - Ben.

We are transmitting CELT encoded packages over a lossy channel, where from time to time complete packages do not arrive at the decoder. My questions: 1. What will be the best pattern to be fed into the decoder in order to minimize the audible impact and make best use of internal states from last block (like overlapp-add history etc.) and best blending with next block ? 2. Does this pattern

Hello, I have a really irritating issue that I haven't had time to investigate much - I hope someone has encountered it and can tell me a solution. I didn't see anything in searching archives / sites.. When my Nextel i90c phone gets a page (2 way text message via the internet option) it has an irritating tone to get me to hear it. However this tone seems to mute asterisk (reproducible).

This is weird. If I have 2 members call into meetme using zap PRI channels on the box, they can here each other's keypresses. If I have 2 members call into a separate box using the same PRI's and then forward (dial(iax/...)) them to the previous box into the same meetme, they only hear a minor "squelch" for each other's keypresses. How can I completely mute a

I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. I'm using the FreePBX web interface, and I can't