similar to: documentation?

I know *8 kind of works with SIP but what about the rest should they work do they work with a zap device? *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid

Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200,

Hi, Lately I''ve been visiting the RoR API site quite often, so I modified a Firefox search plugin to make searches to it (with Google). So here I post it in case someone finds it useful: <SEARCH version = "7.1" name="Google - RoR API" description="RoR API SiteSearch" method="GET"

Hi, I have a function in R that compares two very large strings for about 1 million records. The strings are very large URLs like:- http://query.nytimes.com/gst/sitesearch_selector.html?query=US+Visa+Laws&type=nyt&x=25&y=8. .. or of larger lengths. The data-frame looks like:- id url 1

I'm using omega for a sitesearch and currently having a problem trying to filter $field(sample). At the moment it returns text from the page header and navigation within the sample, ideally I'd like it to return only the page content. I've been trying various omegascript commands to trim the output or seperate the fields and also looking at scriptindex to control how the xhtml is

Hi, Someone knows an Open Source solution that can handle Outbound IVR for asterisk?. What I'm looking it a pre-preprogrammed a telephone call that reach a Person and start making an Interview over the telephone. Specifically I want to call all my customers exactly one hour after the service has been performed and ask some questions in an IVR, also the results of the Interview I will need

Hello everybody, I am making a modified version of "Stephan Steinhaus' benchmark test for number crunching, v. 2, (see http://www.scinetificweb.com/ncrunch/ncrunch.pdf for the original version), comparing several functions of some math/stat software. R is not performing bad at all... except for the sorting of a 1,100,000 random vector (test #3) which is the worst of all (see cell F3 in

Hello everybody, I am making a modified version of "Stephan Steinhaus' benchmark test for number crunching, v. 2, (see http://www.scinetificweb.com/ncrunch/ncrunch.pdf for the original version), comparing several functions of some math/stat software. R is not performing bad at all... except for the sorting of a 1,100,000 random vector (test #3) which is the worst of all (see cell F3 in

This is probably a stupid newbie question. Is there a way to search the list archives? Nick Stein NetPiano.com nick@netpiano.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050317/e8daad09/attachment.htm

I wish to perform brain surgery this afternoon at 4pm and don't know where to start. My background is the history of great statistician sports legends but I am willing to learn. I know there are courses and numerous books on brain surgery but I don't have the time for those. Please direct me to the appropriate HowTos, and be on standby for solving any problem I may encounter while

Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia,

Hi guys! I?m using Asterisk 1.2 with mISDN support. I have problems with Pickup calls with my Grandstream Buttons . I set up on Dial Plan this: Exten => _**XXX,1,Pickup(SIP/{EXTEN:2}) but it doesn?t work if the call comes from mISDN. So, I wanna do something to this: Exten => _**XXX,1,SendDtmf(*8#) because if I introduce *8# into my telephone i can pickup a call from everywhere. BUT the

Hi Michal, I'm cc'ing the NUT development mailing list. you should also subscribe to it. 2010/10/28 Michal Soltys > Greetings > > I have loads of relatively old smart upses that I currently control using a > bit patched version of apcupsd - the reason for those patches is the age of > the upses - roughly 10 years old+, which are can be a bit tricky to control. > From

Hi, I'm trying to put together a simple gateway configuration involving Asterisk. I have a machine with 2 Digium X100P FXO cards installed and the Asterisk Software, and I have 2 Sip Phones defined. What I want to achieve is, any call arriving at FXO 1 is forwarded to Sip phone 1 only, and any call received on FXO 2 is transferred to Sip Phone 2, conversely any call originating from Sip

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

On 6/17/23 08:47, Steve Matzura wrote: > > Both Background() and WaitExten()  allow the caller to enter DTMF > digits. Asterisk then attempts to find an extension in the current > context that matches the digits that the caller entered. If Asterisk > finds a match, it will send the call to that extension. > > > My question then is, is "*" a valid exension, as

Hi All! Have a problem with remote call pickup via sip. When 1 sip phone is ringing and I'm trying to pickup a call from another sip phone by dialing *8# I'm getting: -- Sent into invalid extension '*8#' in context 'from-sip-post' on SIP/ciscok-8d39 such configs: zapata.conf ------ context=inbound-analog callgroup=2 channel=2 ------ sip.conf ------ [ciscok] type=friend

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

Hi, Has anyone used R to send out an email via an SMTP server? I run R from a script and would like to be able to send out an email from there also. I could shell out to an email program, but was hoping maybe someone had a package for R or a smooth way to do it. Thanks, Josh [[alternative HTML version deleted]]

Hi, I asked a time ago what were callgroup and pickup group used for. I have done some proofs and all, and I'm not sure if I have pick the idea up well!! That's what I understand: For example: group=1 callgroup =2 and pickupgroup=2 and my phone is a membership of the group 1. that's mean that when a phone that belong to group 2 is ringing, I'll be able to answer this call dialing