similar to: (no subject)

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

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

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:

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

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/.

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 =

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

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

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

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

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?

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,

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

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

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

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

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

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

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"

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 >>