similar to: sip problems

Displaying 20 results from an estimated 400 matches similar to: "sip problems"

2009 Oct 09
1
Substituting the extracted coefficients into the formula, exctracted from the result of nls()
Dear all, Here I come with another stupid question. Suppose I want to use nls() to fit a series of data (here modelled by generated points), then plot the points and the fitting curve. I figured out some way of doing it: x <- runif(1:20, 0, 10) y <- 0.1*x^2 - rep(3, length(x)) + rnorm(length(x), sd = 0.5) yfit <- nls(y ~ a*x^2 + b*x + c, start = list(a = 1, b = 1, c = 1),
2009 Sep 28
2
Regression line w/ residuals - tuning the plot
Hello everybody, I am sure this is a beginners' problem which is being asked recurrently every few months, but nevertheless I wasn't able to find the answer searching through the r-help list. So here is my problem: I would like to plot a set of points (y vs. x), a (linear) regression line through them, and on the second graph underneath the first one I would want to plot residuals vs. x.
2006 Jan 31
5
Queue() with timeout=0
Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=
2005 Feb 11
2
transferring a IAX call into a conference
When I make a call out on the Faktortel number I am then able to transfer to call to my asterisk meetme room of 801 by hitting 'transfer' then '801' then 'send' on my grandstream phone. This connects my faktortel trunk (and who ever is on the other end) to my conference room I can then make another call using my local pstn service and set up a 3 way (or whatever number
2007 Mar 26
1
Asterisk incoming caller id problem
Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New
2006 Feb 28
1
FW: Re: Delay on Phone ringing
Skipped content of type multipart/alternative-------------- next part -------------- asterisk1*CLI> soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' --
2007 May 15
1
Asterisk is not showing the correct Incomming CallerID
Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the
2009 May 16
1
Queue Load, Asterisk Disconnected
I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for a queue up to 15 agents through a PRI line, it was working fine for more than 1 year, suddenly, when there is a load on the queue, the asterisk service disconnects and the calls are dropped. And the service starts again after few seconds, and so on. I am not using fax. I checked PRI by zttool and there are no alarms. The cdr logs
2005 Sep 01
1
dialparties.agi is returning no extensions to dial
Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if
2006 Feb 27
3
Asterisk with HT 488 FXO
Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:
2006 Jun 09
2
No CID on ZAP
I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP
2009 Oct 08
4
Dialplan problem
Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings
2005 Feb 17
2
arrgghhh dialparties.agi
Hi I've been looking for 10 minutes and cant find dialparties.agi Can anyone tell me what folder this is located in as I'm going crazy. (if it makes a difference I use asterisk@home and am replacing the AMP dialparties.agi file) Super big TIA, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Feb 17
5
chan_capi problem
Hi to all I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus. I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start: [chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group Feb
2006 Mar 30
3
asterisk doesn't wait for whole extension
Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When
2005 Mar 22
1
No recorded messages
I have installed my first Asterisk implementation using the Asterisk@home ISO. I am using the SJPhone software. Using the setup page, I have been able to configure two extensions. Whne I dial from one to the other, the other does not answer even though it is registered. Watching the log in the CLI, I can see that recorded messages are being played;: == No one is available to answer at this time
2012 Jan 12
9
Re: [PATCH] add netconsole support for xen-netfront
On Wed, Jan 11, 2012 at 04:52:36PM +0800, Zhenzhong Duan wrote: > add polling interface to xen-netfront device to support netconsole > Ian, any thoughts on the spinlock changes? > Signed-off-by: Tina.Yang <tina.yang@oracle.com> > Cc: Konrad Rzeszutek Wilk <konrad.wilk@oracle.com> > Cc: Jeremy Fitzhardinge <jeremy@goop.org> > Signed-off-by: Zhenzhong.Duan
2006 May 24
5
macro-dial
Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani
2004 Jan 02
4
Newbie - getting two local phones to communicate would be a good start :)
Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a
2010 Jan 12
1
Multiple symbols per single line in a legend
Hello everybody, Is it possible to coax legend() into displaying more than one simbol per line in legend? I have a graph like the one attached to this mail; I would like to reorganize the legend in such a way that the duplicate text would be omitted, i.e., the first line would read <square> <triangledown> "increasing frequency" and the second one would read <circle>