similar to: force channel hangup

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

I realize this may be less of an Asterisk question and more of a... well... everything but asterisk, but still relating to asterisk question. I was looking for a Click to Dial/Web Dial solution and I found AsteriDex. I'm looking for something I can use on the road where I can hit an internal Click to Dial/Web Dial page from my cell, tap on a number and have it bridge a call between

Hi, I am learning cucumber. I have integrated the Devise with my rails 3 application. My problem is I have to write a test which will create a project. For that I have to make sure that user should be logged in. So I have written the user login feature first which is working fine. After that I have written the creation of project scenario. Whenever I am running that scenario we are getting an

Hi all, I''m in the process of creating rspecs for my helpers. One of the helpers in app/helpers/application_helper.rb looks like this: def page_name @page_name || "Define @page_name in #{controller.controller_name}::#{controller.action_name}" end The rspec is simply: it "should something" do page_name end

Hi, I have a SPA-3000 and would like to use the 911 recipe from http://www.voip-info.org/wiki-Asterisk+tips+911. So I took the simple recipe and modified it slightly: exten => 911,1,ChanIsAvail(SIP/potsoutbound) exten => 911,2,Dial(SIP/potsoutbound/911) exten => 911,3,Hangup() exten => 911,102,SoftHangup(SIP/potsoutbound) exten => 911,103,Wait(1) exten => 911,104,Goto(1) Now,

Ok, I have read most of Agile Web Dev... so I had a question about beautifying URLs using routes.rb. I''m trying to architect a content management system in which a user can create template pages. This way, when a user creates a template page nested within, it displays it as though it''s displaying a folder structure, much like a directory tree (using acts_as_tree). I have a

I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

HI There is a question about the fxo of the zaptel card which is set a number to use as common analog phone. When I use ${CALLERID(num)}to get it's number, it could'n be done. But ${CALLERID(num)} could get the other number of the SIP or IAX . Could you tell me the reason, and how I could get the number of the fxo which is used as a common analog phone? Thanks

Hi, I have an 8-port Grandstream GXW-4008 V1.2A ATA converter with analog phones connected to it. They work fine except for just one "feature" I would like to modify. Somehow, each time the ATA re-registers the SIP clients or each time the device has to be rebooted for maintenance, the phones ring once. This feature can be useful as it notifies the user of the re-registration.

We have an issue where Polycom's lose BLF functionality after a reboot. The only way to fix it is to reboot the Polycoms. Anyone else have this issue? We are using 1.4.18. If I run 'sip show subscriptions' all the subscriptions come back after the restart but the lights on the phones do not work. Any help would be appreciated. -Thermal -------------- next part -------------- An

Ok, dumb question. On a certain LAMP server I am seeing in 'ps auxf' a process called "unit" with no arguments or other path info. It has a fairly low pid, 3041, indicating it might have been started soon after reboot (last week). but ps says it was started yesterday, I don't see it on any of 3 other CentOS machines. It is hard to google for such a generic name. So does

Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081127/b41ca08b/attachment.htm

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir

25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: >> I want to change call files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have following settings done on my Fedora8: Downloaded openh323-v1_19_0_1-src-tar.gz pwlib-v1_11_1-src.tar.gz Extracted them in /root/openh323 and /root/pwlib Exported the following variables: PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH Then I compiled pwlib and it was fine. But in

SECURITY QUESTION & SANITY CHECK: If only my SIP ports and a small range of RTP ports are facing the public internet, what is the method by which an evildoer would be able to do fraudulent long distance on my nickel? Would it REALLY be as simple as guessing the credentials for ANY of my local sip endpoints? Like most people, my local endpoint credentials would be easy to guess: Username

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!