similar to: Setting up ADS in Samba with MIT kerberos mapping/backend

Dear R wizards, Hopeful someone can help me with what I believe is a pretty simple task. I pretty new to R so some (much) of the obvious escapes me. How do I get a distance matrix into heatmap? What do I tell distfun if what I'm trying to map is already an ordered distance matrix? I tried >heatmap(x, distfun=as.dist(x)) where x is the distance matrix but R gave me an error. Thanks in

OK, so this is a really stupid question and should be incredibly simple to do but I can't figure it out. So maybe someone would be so kind as to tell me. I have a vector of zeros and ones. I want to replace all of the zeros with "black" and all of the ones with "gray". That's it. Any help would be appreciated. Thanks, Paul `-:-. ,-;"`-:-. ,-;"`-:-.

?I am running Ubuntu 16.04. I am following this guide: https://wiki2.dovecot.org/HowTo/AntispamWithSieve. At the end when I try to compile the .sieve files, I get this error: sievec(root): Fatal: Plugin 'sieve_imapsieve' not found from directory /usr/lib/dovecot/modules/sieve.? Does anyone know about this? I have dovecot-sieve installed and I'm using Postfix and Spamassassin. --

Can anybody tell me where to find a copy of heatmap2? I've seen it in my travels across the web but didn't bookmark it and can't find it again. Thanks in advance. Paul `-:-. ,-;"`-:-. ,-;"`-:-. ,-;"`-:-. ,-;"`-:-. ,-;"`-:-. `=`,'=/ `=`,'=/ `=`,'=/ `=`,'=/ `=`,'=/ `=` >==/ >==/

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

Hi All, Is there any way to connect (register, initiate and receive calls) with Asterisk to FWD through NAT? Since I own my router port forwarding is not a problem. I tried with Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com but since Asterisk still use internal IP in some SIP fields I got "479 We don't accept private IP contacts. Please set your external

Still getting the same error. Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request) please advise anyone!!!!!someone!!! jai

FWD (Free World Dialup) allows any SIP call to US toll free numbers via * 18xxzzzyyyy at fwd.pulver.com This works WITHOUT the need to be registered at FWD so in my dialplan I have something like: exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r) exten => _8.,2,Hangup And I just dial 8-1-8xxyyyzzzz and presto ... calls go through just fine 99% of the time. I'm wondering if

I've been having issues getting FWD to work. I posted this same Q to the FWD forum (no responses yet), but I was hoping someone here had some insight: http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0 I just signed up for an FWD account (I know I had one before, but I lost it.. :) I've got it running through Asterisk - all working fine

I've tried to register * at FWD but * segfaults, so i guess my register-line in sip.conf isn't correct. It looks like: register => fwd#:pwd@192.246.69.223 should it be different? Chris

Hi, I have some doubts on sip.conf. 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? 2) Why we need a section in the sip.conf for the proxy, say, Free World Dialup's fwd.pulver.com? In the case of 1), how to assign the value to section [fwd.pulver.com], since there are more