similar to: How to number extensions - Which way is best?

Hi, I need to run a PCoA (PCO) for a data set wich has both positive and negative values for variables. I  could not find any distancecoefficient other than euclidean distace running for the data set. Are there any other coefficient works with negtive values.Also I cannot get summary out put (the eigen values) for PCO as for PCA.   Thanks. Dilshan [[alternative HTML version deleted]]

When I set up my Asterisk box at home I didn't want to have to dial 9 to dial off premises, so I gave all my local phones three digit extensions with this format: 1[1,0]*. My thought is that there are no area codes that start with 0 or 1, so if I use those numbers, I can create 20 local extensions that can be dialed with 3 digits, and not have to use a timeout when dialing long distance. If

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ

We are trying to implement "area-code dialing" in our asterisk PBX. Basically: we will have a number of customers, who may be in different area codes, that want to direct-dial each other's extensions. We want this to work like a "real" centrex, in that seven-digit numbers should try (1) "local" VoIP extensions, and then (2) "local" PSTN numbers.

Hi, We ran into a little problem recently with our phone provider (Telus Canada): we are unable to dial numbers outside North America. This is what happens: the phone number 011... is sent out over our T1, Telus sees the correct number on their switch. However the switch thinks it's a North American phone number (and thus has to have 10 digits) and rejects the overseas number as

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL:

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

What would be my best option to receive calls via VOIP. I would like to use it as an alternative number when my main number is busy. The solution is not that easy as in order for customer to be a free call DID=Direct Inward Dialing provider would need to be a local company, I think. Correct my anybody if I'm wrong. I'm located in Alberta Canada so my chases are even smaller. I've

Hi... I've been fighting this for a while now, trying clean builds of Asterisk 1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today. No workee. :-( Here's the results for various calls made off-hook (push the blue Speakerphone button on the Polycom 430): 988852700 - Phone waits for me to either hit the soft-key "Send" or "EndCall". If I hit "Send",

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

Residential Long Distance. One of my biggest pushes towards a VoIP provider was cheap long distance. Now in the U.S. at least with SBC they now have a plan for Unlimited Long Distance. The price is 30.00 a month if you do not have a couple of required features on the line already like Caller ID and a feature like 3 Way or Call Waiting etc. in which case that lowers the price to 20.00 a month.

Dear all, I'm analyzing a dataset (A) of 400 cases with 11 binary variables. Unfortunately, several (actually a lot) of cases are identical. NA are also present. I want to to plot distances between cases. For this, I obtained a distance matrix by dist(A, method="binary"). I then analyzed the obtained distance via Principal coordinate analysis with cmdscale(). Results are fine.

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hi Right now I have my url showing as the following: http://localhost/categories/italian/recipes/special-pasta-sauce I would like it to be: http://localhost/categories/italian/special-pasta-sauce (or even http://localhost/italian/special-pasta-sauce) my category model has_many :recipes and my recipes model belongs_to :category (for the purposes of my test project a recipe can only belong to one

Hi, I''m trying to use the SaltedHashLoginGenerator as documented on http://wiki.rubyonrails.com/rails/pages/SaltedHashLoginGenerator Ruby version: ruby 1.8.4 (2005-12-24) [i386-openbsd3.9] I''ve the following commands: gem install salted_login_generator gem install localization_generator rails myapp cd myapp ruby script/generate salted_login User Localization All ran

I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked.... Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten => 107,1,Dial(SIP/102&SIP/107,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup exten => 107,102,Voicemail(b102@pstn)

I''ve installed the puppet labs lvm module. After fixing the missing "}" at line 20 in init.pp (really?!?!), I''m getting this:'' err: Could not retrieve catalog from remote server: Error 400 on SERVER: Puppet::Parser::AST::Resource failed with error ArgumentError: Invalid resource type physical_volume at /truth/sauce/env/prod/modules/role/manifests/common.pp:43

After over a month (well, ok, no more than an hour a day :) of planning, getting hardware, tinkering and testing, I'm about to my Ultimate Home Phone System (tm) online. Connectivity to the outside world is provided by: A. 1 POTS phone line connected through an X100P ($11/month, needed to carry DSL) B. 1 Vonage ATA186 connected through an X100P (needed for the rate center :( ) C. 3 Broadvoice

In the process of turning up a new pri. Zttool indicates the T1 is ready with no alarms. asterisk*CLI> pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3