similar to: Interesting problem with IP's

I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxxxxxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it "forget" what it has programmed and

Hi! I have a 3d array: > dim(ib5km15.dbc) [1] 190 241 19 and a set of positions to extract: > ib5km.lincol.random[1:3,] [,1] [,2] [1,] 78 70 [2,] 29 213 [3,] 180 22 Geting the values of a 2D array for that set of positions would be: > ima <- ib5km15.dbc[,,1] > ima[ib5km.lincol.random[1:10,]] but don't find the way for the case of the 3D array: >

I think I must be missing something obvious, but I'm having trouble getting a data transformation to work on groupings of data within a data frame (csss3) as defined by 2 factors (population, locid). The data are sorted by year within locid within population and I want to lag another variable (dbc), i.e, shift them down by 1 row replacing the first row with NA, within groups defined by

Ok so I see the problem that I'm having creating a new variable (LAG1DBC) in the example data transformation below is that tapply() is creating a list that is not dimensionally consistent with the data frame (data). So how do I go from the list output of tapply() to create a dimensionally consistent vector that can create the new variable in my original data frame? I've been trying

I have a couple of 64-Bit Xeon's with CentOS 4.2 x86_64 installed on them. They don't all have the same motherboards. I've having a difficult time getting CentOS 4.2 x86_64 to boot on some of them in SMP mode (UP works fine). After the installation of CD, it boots up but kernel panics with the following error: Badness in i8042_panic_blink at drivers/input/serio/i8042.c:992 This is

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A

Hello, Did anyone succeeded in installing Asterisk on OpenWRT system. pls help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090724/aee7ee12/attachment.htm

Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a

I tried implementing my * and it didn't pass the spouse factor at this time. I wanted to hook it up for outbound only at this point to get a better handle on the dial plans and the echo problem. I thought this might have been done before as a natural part of testing - but maybe not. In wcfxo.c I found this: if (!wc->offhook && !wc->ringdebounce) { if

Cisco 7960 has six buttons/lines. Can some of them be configured for fast dialing? If it can't be configured on the phone, how can I configure it on Asterisk? -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr

I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the

I cannot reach voip-info - is it just me or is the site not available ? Julian

Hello, I'm writing a set of python scripts that need to consult samba ldap database without authentication. I've set up 2 machines, the first as PDC, the second as BDC. I run my scripts on both and while they run fine on the PDC they end with error on the BDC. Here is the minimal example that behave this way: import ldap LDAP_URI =

I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get

I experience random crash of machine (full hang, requiring a hard reset) after trying to test run Asterisk 11. The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installed Anyone experience similar situation? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?