Displaying 20 results from an estimated 200 matches similar to: "(no subject)"
2004 May 28
1
Zap callgroup/pickupgroup question
I'm trying to set up asterisk so any phone connected to channel 1-16 of my
Adit600 channel bank can pick up a call coming in on channel 24. I do not
wish to ring any of the 16 channels on an incoming call -- this is strictly
so I can pick up the line if I see it ringing and wish to answer at work.
I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3.
However
2005 Mar 13
4
SUSE 9.2 and Zaptel channels
Of course I am not a kernel expert, so .. please be patient.
I am investigating on my zaptel/zapata problem.
As the main error message asterisk quits on mentions <'/dev/zap/channel':
No such file or directory> I went peeking over there.
[Asterisk Verbose Error
Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/
dev/zap/channel': No such file or
2004 Jun 01
1
Zap and call pickup -- it don't work.
The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24
FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a
call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16.
*CLI> show version
Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux
The zapata.conf and extensions.conf are located here:
2011 Sep 13
12
Assertions for asynchronous behaviour
Hi all,
In GOOS[1] they use an assertion called assertEventually which samples the system for a success state until a certain timeout has elapsed. This allows you to synchronise the tests with asynchronous code.
Do we have an equivalent of that in the Ruby / RSpec world already? I know capybara has wait_until { } but that''s fairly rudimentary - the failure message isn''t very
2012 Apr 04
5
Simple code dosn't work
I think you have a syntactical error on the line thats throwing the
error, you state:
> j.even?should be true #throws an error on j == 2, j == 4
should this line not read as:
j.even?.should be true
--
Posted via http://www.ruby-forum.com/.
2005 Jan 17
1
Attempting native bridge
ERROR CONDITION
---------------
-- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered SIP/2001-f6c4
-- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead
Have searched web and archive w/o good results.
Thks in advance for any help,
Dave
sip.conf
--------
[general]
port =
2004 Jun 11
1
"Caller ID" question
Since caller ID does not work with my FXO card, I am wondering if Asterisk
supports the following extensions functionality.
When a call comes in, I'd like to give the caller an opportunity to enter an
extension if he/she knows it; if not, Asterisk will dial one or more default
handsets. I know Asterisk can do this, but is it possible to change the
default "caller id" when the call
2004 May 12
3
Cisco 7960 SIP - DND soft key toggle?
Running the latest * CVS and Cisco 7960G and 7940G phones with SIP 6.3 image.
I have figured out how to turn on the DND feature through the
Settings>Call Preferences>Do Not Disturb - Yes then Save. This puts the
phone into DND On and shows a DND image above the far right soft key which
you use to turn off DND.
There should be a better way. An on/off toggle of the soft key that it
2020 Aug 08
2
My first real submission with Phabricator
Madhur Amilkanthwar via llvm-dev <llvm-dev at lists.llvm.org>於 2020年8月9日
週日,上午1:53寫道:
> Hi Paul,
> I hope you have gone through
> https://llvm.org/docs/Contributing.html#how-to-submit-a-patch.
>
> Generally, I would do 'git add' on the new file. 'git diff' should show me
> the newly added file. Further, I'd just do 'arc diff' and this should
2004 May 06
4
Playing GSM files in Windows
For the archives...
In trying to play GSM files in Windows (Windows XP for me, but in
general) I found no help on Google, so when I figured it out I thought I
would post it here.
Q: How do I play GSM Files in Windows?
A: Use Quicktime, it supports the GSM audio format directly.
Andy Farnsworth
farnsaw@stonedoor.com
2020 Aug 08
3
My first real submission with Phabricator
I am ready to submit my first real submission for review with Phabricator. Please forgive my meager knowledge of Git. I did a 'git diff' to generate the diff file. The contents look good. However, there is one new file, a TableGen test file. How do I get that file included in the diff, or otherwise included in the submission?
2008 Jun 27
1
Asterisk, POTS and plain handsets
Hello,
I've spent a couple days searching and posted into the forum with no luck, apologies
to anyone who reads the Digium forums for the cross-post.
I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS
line. Also connected to the POTS line are plain telephones, non SIP, just plain
old telephones. When one of the normal handsets goes off-hook,
2004 Jul 19
4
TDM400P Internal Extenion Config
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
I have a TDM400P with one FXO module and a FXS module. The main problem
I have is not being able to get the extension attached to the FXS module
to ring or be able to make calls. It gets a dialtone fine but I
2003 Nov 04
2
asterisk does not hang up
hi,
i am trying to do to autoattendant. here is my
extension.conf part
[tumpak]
exten=>s,1,Dial,Zap/4|10
exten=>s,2,Voicemail,u9999
exten=>s,102,Voicemail,b9999
exten=>t,1,hangup
so when a caller dials the extension 2 suppose, it
enters to the above context.. everything is fine. the
problem is when the caller hangs up the asterisk does
not. after caller hangs up and tries again he
2004 Sep 27
1
Fedora2 and zaptel - using the udev
Hi,
I am sorry if this message has been reposted, but for some reason I am
having problems with posting it.
I configured asterisk and zaptel modules with fedora2.
I want to be able to load the zaptel wcfxo and wcfxs modules.
For now I will use only the Wildcard TDM400P card.
I am able to load the modules but I cant configure them using ztcfg or
zttool because the tools are compiled to use the
2005 Jan 05
5
"Out the box" solutions?
Hi, again.
I've spent a week trying to get asterisk to work on FreeBSD unix, with some
success. Everything works until I plug the box into the TELCO line and then
the line goes off-hook and stays that way.
So I bit the bullet and decided to install the application on a fresh linux
install. Not to start an OS war, here, but linux is ... difficult ... for an
old unix hand to get his mind
2003 Apr 24
3
Collecting dialed digits
I am trying to set up an auto attendant for the first time, and am having
trouble getting to the submenu. My extensions.conf file looks like this:
[incoming]
exten=> s,1,Background,menu1
exten=> s,2,Wait,20
exten=> s,3,Goto,s|1
exten=> 1,1,Playback,option1
exten=> 2,1,Playback,option2
exten=> 3,1,Playback,option3
It is my understanding that asterisk treats the digits entered
2005 Feb 08
2
Voicemail not working properly
i am working on asterisk. i am using fedora core 2 on
my asterisk mechine. when i was working on stable
version my voicemailmenu was working well. i can
lissten to menu and send dtmf to control menu now i
have compiled CVS version of asterisk. now when i
configure my voicemail for any extension suppose i
declared a voicemail box 9999 for user 3000. when i
dial to 3000 i cannot have any menu there
2011 Dec 23
6
http session nil
hi all,
i am new in ruby on rails,i have one issue regarding the session in ruby
on rails.I am using session , i have following condition in my rhtml
<%if @session[''user''].first_name == "admin"%>
<table width="290" border="0" height="20" align="right"
cellspacing="0"
2017 Apr 02
4
sound problems... alsa & systemd?
On 03/29/2017 06:43 AM, ken wrote:
> On 03/28/2017 08:53 PM, ken wrote:
>> The www has failed me with this, so I'm trying you guys. Sound worked
>> great out of the box when I installed 7.2... Yay! I could watch all
>> kinds of videos, like on facebook and youtube. And I could listen to
>> most podcasts too. But then something happened. It was either a
>>