similar to: Can you caculate with me?

may some one please help me to sort this out, i am trying to writ a R code for calculating the frequencies of the amino acids in 9 different sequences, i want the code to read the sequence from external text file, i used the following code to do so: x<-read.table("sequence.txt",header=FALSE) then i defined an array for 20 amino acids as following:

I have ASTCC installed, and compare it with NuFone, however, I find that the billing of NuFone is always a few secondes more (6 to 24 seconds) Does anybody has an explanation / solution for it? bye Ronald

Hi,All I use HTB to control traffic.Here is my script: /usr/sbin/tc qdisc del dev nas0 root /usr/sbin/tc qdisc add dev nas0 root handle 1:0 htb default 60 /usr/sbin/tc class add dev nas0 parent 1:0 classid 1:1 htb ceil 896kbit rate 896kbit prio 0 burst 60k /usr/sbin/tc class add dev nas0 parent 1:1 classid 1:10 htb ceil 896kbit rate 537kbit prio 1 burst 60k /usr/sbin/tc class add dev

Can anybody help me and explain me the following area codes: 1-340 US-USVI 1-670 US-CNMI 1-710 US-Governement 1-787 US-Poerto Rico 1-802 ~ 1-808 ??? 1-939 US-Poerto Rico 1-600 Canada Are the above codes are USA Continental tarrif (NuFone / Broadvoice ... )???? What are the codes for mobile phones in USA? bye Ronald

Anyone know how to obtain firmware and starter .cfg files for Polycom phones? Despite registering at the Polycom web site, I can't locate this stuff.

Dear R Users I'm writing my first simulation in R. I've put across my problems with a smaller example in the attachment along with the questions. Please help. Best regards Meenu -------------- next part -------------- mydat<-read.table(textConnection("Level spread change State 4.57 1.6 BlF NA 4.45 2.04 BrS NA 3.07 2.49 BlS NA 3.26 -0.26 BlF NA 2.80 0.22 BrF NA 3.22 2.5 BrS NA

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

anyone else have the above issue? this is today's CVS. thanks.

I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger

In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single line anyway, I wonder if there is a way to check if this (provider) line is taken already. How can I do that? Same is with the phone. How can I see in CLI if a phone is now in use or not? "Sip show peers" shows me just if it is

I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway!!!! I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger

*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * *0 Dial option is tTwWr I tried to call from 601 to 615 601 keys in *0

Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of

I used: cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds iaxyprov astcc and in the same order I try to compile it. Asterisk ends with the lines below. It complains of a newer libpri, but I just did it a step before! What do I miss? chan_zap.c:62:2: #error "You need newer libpri" chan_zap.c:128: error: parse error before '<<' token chan_zap.c:133:1: