similar to: Interesting VoIP device

Hi, I hope some one can help. I need to compute Thurston's case 5 on a large set of data. I have gotten as far as computing the proportional preference matrix but the next math is beyond me. Here us my matrix 0.500 0.472 0.486 0.587 0.366 0.483 0.496 0.434 0.528 0.500 0.708 0.578 0.633 0.554 0.395 0.620 0.514 0.292 0.500 0.370 0.557 0.580 0.615 0.329 0.413 0.422 0.630 0.500 0.783 0.641 0.731

Hello Chuck! I have 128kbit/s for 70 computers and if several users start several FTP/HTTP/Torrent downloads (or one downloads with several threads) or also open several htmlpages with big content, for other users remain not very much. As i hear Squid make traffic shaping on IPaddress base and in my scenario every user will work in equal borders (as another) for all his connections.

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file

For those of you that have dlink VoIP hardware, I thought I'd post this link. It's a whole lot easier that navigating their support site trying to find anything. There is updated firmware for most of their products. ftp://ftp.dlink.com/VoIP/ A word of warning about the DG_104S firmware. It's better than the firmware that came with my box, but it still has quirks. Some settings

Just a quick query to find out if there are any other Asterisk users in Colorado. If you're out there, drop me a line off-list. I'd like to start a user group if there is anyone else out there. --Karl -- Karl Putland <karl@putland.linux-site.net>

I was fooling around with echo cans today. I found that I could reproduce this problem regularly t400p adit600 for fxs mec3 enabled Two speaker phones side by side within about 8" of each other. Put phone 1 offhook with speakerphone Dial phone 2 Phone 2 rings and is loud enough to be picked up by the mic in phone 1 By the second ring the echocan thinks it's figured things out, but the

Following is a log from an attempt to record and playback a file in wav49 format from an AGI script. COMMAND: stream file aa/after_the_tone "" 0 RESULT_LINE: 200 result=0 endpos=41920 RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')} COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 "#" 20000 0

ok all. I have sent the PHP web gui code to Mark at Digium but have not heard back yet. I dont know that status if he wants to CVS it or not. maybe if he does not answer in a this next week ill just upload it to a website for all to DL. we could add to this as a base and get it to work better for all again the current URL for it is http://rads.netcom.utah.edu/openconf/openconf.php Dave P

Is there an R function that I can use to calculate the p-value from the Z statistics computed for the relationship between chance and observed proportions in predictions. More sprcifically I am refering to proportional chance criteria (Cpro). Details are in Huberty's book on Applied discriminant analysis, unfortunately our library has misplaced the book. I got some details from the

I''m trying to use Canoo WebTest (based on HtmlUnit) to test my webapp after integrating scriptaculous. While my test passed, I get a nice long exception message which barely makes any sense (see below). I found I could get rid of this message by commenting out line 114 of effects.js: 114 this.timeout = setTimeout(this.loop.bind(this), 10); Any idea of what might be causing

On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: >> Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. >> >> So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do?

Skipped content of type multipart/alternative-------------- next part -------------- Index: pbx.c =================================================================== RCS file: /usr/cvsroot/asterisk/pbx.c,v retrieving revision 1.14 diff -u -r1.14 pbx.c --- pbx.c 19 Apr 2003 02:41:22 -0000 1.14 +++ pbx.c 21 Apr 2003 02:27:43 -0000 @@ -713,6 +713,8 @@ { char *first,*second; char tmpvar[80] =

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

Hello again list, I'm running a 1.4.42 install on SUSE with an AEX410P card. The DAHDI release is 2.4.0 because the machine won't properly install 2.5 and also won't install Asterisk 10.0 because I can't get a good SQLite3 library to install. Whenever I enter DTMF very quickly or very slowly, app_read des on me. Has anyone experienced similar joy

Phil Wyett wrote: > On Wed, 2016-03-23 at 09:20 -0400, m.roth at 5-cent.us wrote: >> Now, this is one we have an issue with: it's got a bttv card, and motion >> running on it. It's a Dell PE R720. For some reason, it has never liked >> the card: 20 min after reboot, it says fatal bus error... but nothing's >> wrong, and it runs just fine. Well, expect that we

Hi, It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem.. My setup.. UA1--[AST1]--{IAX}--[AST2]--UA2 | | PSTN1 PSTN2 I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent.. I

Bill I have adtran750 with Vista/Aastra 350, 390, & 480's never seen that one tim -----Original Message----- From: asterisk at billheckel.com <asterisk at billheckel.com> To: asterisk-users at lists.digium.com <asterisk-users at lists.digium.com> Date: February 21, 2003 12:52 PM Subject: [Asterisk-Users] Fun new problem with Aastra 390 ADSI phones >I seem to have

We want to test capacity of our MeetMe room. The thing that is distinct about this is that the incoming line is being delivered IAX2 to our server across the net - so Telephone -> VoIP Gateway -> MeetMe. We want to test both the VoIP Gateway and the MeetMe room performance. You can reach our MeetMe room directly at 1-301-561-9229 If you want to test with us we're thinking maybe 9pm