similar to: Policies - deny some nubers

Hello, how is posibble to search only parts of words for exapmple if i have in index "Lazy fog was jumping over microstadio." I want this one will return for following search: "micro" -- Posted via http://www.ruby-forum.com/.

Hi, today we have a problem our mail server about user accounts. (something about prefetch) anyway i corrected, but some clients re-download emails again. how can i stop this .I've read about some forums about UIDVALIDITY ? is it posibble prevent clients redownload emails with pop3 ? thanks in advance. -- Sel?uk YAZAR

hi all, i want to ask about caller id in asterisk. if i make outgoing call, people who receive my call can see the pilot number. what i must do if i want that people receive my other number (DID number)? is it posibble? thank to everybody.. regards, shahdan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

hello what is the differences between wine and vmware? thanks in advance.

R-users, Attached is the file (SR_use2.txt) I'd like to include and includes column headers. nat_est is the response variable and is the number of species at a particular point. The other variables are the explanatory vars (vark, var2, var1, UK, U2, U1, GK, G2, G1, PK, P2, P1). Here is Walsh's sample code for hier.part: data(urbanwq) env <- urbanwq[,2,8] hier.part(urbanwq$lec,

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

Hello All, one of the extensions fall into a loop, I don't know how to hangup that channel -- Executing [i at autoatten:2] Goto("Local/100 at sipphones-000001b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200 at sipphones-000001b2;2 -- Executing [i at autoatten:1]

Hi all, I have a data frame with a variable Description containing text of speeches and I would like to count number of sentences in each speech, > str(data) 'data.frame': 255 obs. of 3 variables: $ Group : Factor w/ 255 levels "AlzheimerGroup1","AlzheimerGroup10",..: 1 112 179 190 201 212 223 234 245 2 ... $ Gender : int 1 1 0 0 0 0 0 1 0 0 ... $

Hi, I got this setup. analog phone (ext7) ---> analog pbx ----- (ext 6 analog) voicemodem (ext 3 asterisk) ---- ttyS0/asterisk ---- sipphones q1: I got the voicemodem to work, but oneway only. I can talk from my analog phone, to my sipphone, but not the other way ? I know it only suppose to works in half duplex, but nothing come TO the phone. q2:

Hi, I have the following problem. I have 5 phones behind the same nat (canreinvite=yes). it works fine to receive calls and to make calls. sound quality is good, so everything works fine. The poblem is that the phone behind nat cant call each other. It works if canreinvite=no. But i want to do this. Does anyone have an idea? Regards, cjk.

Samba 4.0.4 installed, provisioned by classicupgrade, running on Debian Squeeze: -------------------------------------------------------------------------------- The issue is, that changes to the roaming profile is not transferred after log ins/outs between Win2K and XP machine. In example: I log into the W2k machine with my testuser and create a "testdir1" and "testdir2" on

> net:\\server > Sometimes, you have to read documentation. Oh, no. It have to do users, who use Far. :) I remembered just first camt to my head. :)) And try to explain it of nubers of users. Especially acconuters... You as Russians can know it. I know some other soft where there is no such ability (access by \\server\folder) But can not remember it at the time. > That's simply not

If my messages seems somehow unreadable - I sent it from Gmail Web UI. mathias dufresne, read my 2 or 3 last messages. I wrote about mounting \\server\share as disk and risky fo viruses crypting files. Also read messages other, who does not work in AirBus. And more over, it' s your oppinion. But I see useless of more discussion at atll. If you want to combine your efforts to help with

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104 at sipphones:1] MeetMe("SIP/101-00000813", "104") in new stack == Parsing

Hi All, I recently upgraded from a very old stable to HEAD. For some reason, incoming callers don't hear ring tones when calling in. Everything else is working fine. Where should I look for a fix? ISDN --> E100P --> asterisk --> sipphones. Thanks Johan

Hi all, I'm having issues with overwrite caller id, when I call someone my caller id should be "mycompanyinc" but instead my id shows up as my extension number 101. this is what i have in sip.conf [101] type=friend context=sipphones call-limit=99 callerid="iuser 101" disallow=all allow=ulaw allow=alaw username=101 secret=Passwd dtmfmode=rfc2833 host=dynamic mailbox=101 at

Hello, I want to share mailbox between two extensions Ext. 101 Ext. 102 I want the messages to go to mailbox 101, when when checked mailbox from extension 102 to be able to clear the bliking red light. here is extensions.conf exten => 102,hint,SIP/${EXTEN} exten => 102,1,Dial(SIP/101&SIP/102,20,t) exten => 102,2,Voicemail(101,u) exten => 102,102,Voicemail(101,b) exten =>

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

www.spacialaudio.com - take a look at SAM.. Bryan ----- Original Message ----- From: "Rick Wilson" <colly@undernetlinux.com> To: <icecast@xiph.org> Sent: Tuesday, August 14, 2001 8:21 PM Subject: Re: [icecast] install instructions for iceS on i.cantcode.com > > Hey ya all , > > I am just wondering , if anyone knows of WebSite scripts for ICECAST < that >

I have an idea, tell me if this wouldn't work... I know it's really ugly, but it might help some people until we can get round robin DNS checks for peers... Since * does not do GetHostByName() again until you reload your config, and BroadVoice and I'm sure other sip providers are using round robin DNS, why not create 2 [<your server here>-out] contexts in sip.conf, and then in