search for: fullerton

To the volunteers of R-Help. Hello, I am currently stuck on an RStudio assignment. The assignment involves creating a double bar graph with the provided info http://math.fullerton.edu/mori/data/introstats/pennstate3.txt My professor has only gone over the very basics of RStudio and we only learned how to make a simple bar graph and labeling x and y axis. The specific directions from my professor: Draw a side-by-side bar graph of the variables WtFeel and Sex. The x-axis, sh...

...1,Answer exten => s,2,Wait(1) exten => s,3,Set(TIMEOUT(response)=10) exten => s,4,Background(enter-ext-of-person) exten => i,1,Playback(invalid) exten => i,2,Goto(s,4) exten => i,3,Hangup exten => t,1,Playback(goodbye) exten => t,2,Hangup include => tests include => fullertonpbx include => intercompany Thank you for any insight you can provide. Dave Fullerton

Hi all, I''m a software developer from New Zealand. I''ve built a couple of rails sites and have some ajax experience. I''d like to develop on a rails site with the aim of making a business - is anyone else looking to do the same? The two rails sites I built recently are foopad.com and friendr.com. Regards, Ben

Not sure if Polycom is changing their policy or if this is an accident, but you can actually download SIP 3.1.1 right from their web site. Anyone looking for firmware should get it now before it disappears. SIP app and release notes can be found here: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html -Dave

On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey All, AT&T is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have

Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed

Since we've had discussion about DST on polycom I thought I'd pass along the rule I used to configure DST on my sipura units as well (This way the date and time passed in caller ID will be correct). Under the admin view go to the regional tab. At the bottom under miscellaneous enter this in "Daylight Saving Time Rule:" start=3/8/7/2:0:0;end=11/1/7/2:0:0;save=1 This is based

honored by DSCP (first 5 bits)- even old equip should be DSCP "compatible"...or I need to do more reading :) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, October 01, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] QOS/DSCP for IAX? Michelle Dupuis wrote: > I actually see the TOS setting in iax.conf, but the default (commented > out) is EF - which doesn't even match a valid bit combination > accordin...

In RELENG_4_8 /usr/src/UPDATING, I see: 20030924: p9 FreeBSD-SA-03:15.openssh Fix PAM-related bugs in OpenSSH's challenge/response code. But there's no mention of FreeBSD-SA-03:15.openssh on this list, the security-notifications list, the web site, the ftp site, etc. Is this advisory still pending? or is UPDATING just mislabled? Thanks, Bryan

Hello all, I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I have asterisk set up to connect to my Dovecot IMAP server and I can leave and retrieve messages from my inbox and old messages. However, I am unable to move messages between folders. I get a message from asterisk stating "Sorry the users mailbox can't accept more messages". Here is my setup: In

Hello all, I have two machines I'm connecting with TDMOE (dahdi dynamic spans) and I have a question about timing parameters. By my understanding one machine should be the source of the timing and the other a slave of that timing. So on machine A I have the following in system.conf: dynamic=eth,eth0/00:0C:29:55:89:7E,24,0 On machine B I have this is system.conf:

Greetings. I have a scenario I would like some advice on. I have a 100,000 square foot building that we will be moving some work crews into. It has offices on each end of the building and a fiber line between them. I currently have an asterisk 1.2 system in place and about 30 phones. My problem is they want a few fax machines out in the warehouse area where I currently have no wiring for

Not sure how I should officially report this, but I'm getting a compile error with DAHDI-linux 2.7 when I define CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h. I am able to compile successfully when I leave it undefined, but I need to be able to use the network support. <snipped> /oct6100_api/oct6100_tsst.o AR /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/oct612x/lib.a

...notify me: Mike Fleming Co-Chairman, Statistical Computing Section Washington Statistical Society mfleming at nass.usda.gov 703-235-5213 ext. 170 The answers to my questions will help me in preparing for my talk: 1. I saw a reference to the Fullerton Library in one of the r-help messages recently. What and where is it? 2. Has there been developed a function like sas.get for R or can the one found at statlib in the directory for S be used? 3. Shell commands like !pwd do not work on my installati...

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

Hello everybody I am trying to connect my asterisk to a payment equipment trough PSTN. I have a TDM400P card with an fxs module an the equipment use modem to send data! I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? Thanks,

Hi, Say I wanted to know what context a SIP registration is using to dial out in my dialplan, what would I do? For example, I have phones on a "local-calls-only" context (as defined in sip.conf), others in "unrestricted-calls". In my dialplan, I`d like to act on that knowledge. Mike -------------- next part -------------- An HTML attachment was

...own[192.168.212.227] Apr 28 13:42:14 mail3 postfix/smtpd[21859]: disconnect from unknown[192.168.212.227] ehlo=2 starttls=1 commands=3 I keep getting smtp timed out, it takes a while, but does time out. _*Using openssl s_client -connect 192.168.0.242:25 -starttls smtp*_ subject=/C=US/ST=CA/L=Fullerton/O=xxxx Law Group/CN=mail.xxxxlawgroup.com/emailAddress=postmaster at xxxxlawgroup.com issuer=/C=US/ST=CA/L=Fullerton/O=xxxx Law Group/CN=mail.xxxxlawgroup.com/emailAddress=postmaster at xxxxlawgroup.com --- No client certificate CA names sent Peer signing digest: SHA512 Server Temp Key: ECDH,...