similar to: app_changrab.c released on pbxfreeware.org

Hi all, there is an application called changrab with quite interesting capabilities: http://www.freeswitch.org/asterisk_stuff/app_changrab.c I think it is available in the 1.4 version by default!? This application can connect to channels which are already UP. The only possibility AFAIK to connect channels after they are UP are the well known conferencing applications meetme and conference. If

When I look at the lastest UniMRCP manual, they only mention as high as Asterisk 13. Does anybody know if I need to do anything to allow it to work on Asterisk 14 and, if so, what that is?

Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! Please

Sorry, this is off topic to asterisk itself, but is about the list server. I had a power failure lastnight at home, where my email server resides, and my network was down for about 20 minutes, that was after 45 minutes of uptime on UPS. Since power was restored, around 9:45 PM EST on 2/16, I have not received a single post from the users, biz, or dev lists. Normally when this has happened

Hello all. I am new to the list and relitively new to FreeBSD. I currently have a server running 4.8 as a dedicated server with cPanel added as a way to speed up the creation of sites and such on the server. I host only a couple of site because I do this in my spare time and don't know enough to be a paid participant in the hosting community. Anyway, on to the question, lastnight, the server

August 5th, 2005: Craig Southeren announced today that OPAL (http://www.voxgratia.org) now provides support for the IAX2 protocol(Written by Derek Smithies and released under the MPL). This support allows you to use chan_woomera (http://www.pbxfreeware.org) driver developed by Anthony Minessale II to interconnect your asterisk systems and use the IAX2, SIP, and H.323 protocols. I would

I am in the process of authoring a Samba-LDAP-PDC HOWTO for Mandrake users that will allow cross platform ( Linux / M$ ) network user authentication. Lastnight I ran rpm --rebuild --with ldap samba-2.2.7a-2mdk.src.rpm and it produced a list of packages that I needed to install in order to compile this version of samba. I did so but I neglected to write down the list. I did the install of

I use Asterisk in my house. Each phone is a different extension. I really like the ability to have multiple simultaneous calls in the house. However, I do miss being able to be able to pick up a phone in a different room. Currently, I have to either transfer the call or transfer the call to a "conference" extension to move around the house. While a connection in progress on one

Hello All I just recently adopted a server with Gentoo/Xen as domain0 server with several domainUs attached to it. Please forgive my xen-noobness, but I was hoping that someone can help me out. These images have not been updated in a long time, and I''m trying to get all the domainUs updated with their appropriate services patches, etc. I''m currently working on one Gentoo

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

Hi all, is where a possibility for simply parsing and changing variables for bad characters ? eg. removing a '/' from a number dialed by a manager-connected application changing 123/4567890 to 1234567890 via bash you could simply use 'echo ${exten/\//}' but i couldn't find a working solution for the asterisk-extensions.conf best regards Dirk Rieger Diese E-Mail und

Currently at our office, if I want someone else to pick up a call, I have to transfer the call to them. So I'm looking into call parking, which is ALMOST perfect. The missing piece of the puzzle: I'm extension 203. I want any call I park to get parked at extension 2203. I want a call my boss parks to park at 2205, since he's ext. 205. In other words, I want calls parked FROM

>1- Call comes in without callerid >2- AGI script answers line >3- AGI script asks to record name >4- Park the call and get the parked extension number >5- Ring all the phones in the house (exec Dial) >6- If phone is picked up, play recorded name >7- Wait for DTMF to accept or decline call >8- If accepted, bridge parked call and current call. Mike, I am wanting this

Folks, I saw David''s tutorial yesterday at RailsConf and started working on RSpec lastnight and am having a *ton* of trouble with the bundle for TextMate. I''ve made *some* progress, but thought I''d throw it to the list * I run ruby, including rb-rubygems using the MacPorts install (per JDD''s Sandboxing Rails) * installed rspec 0.9.4 (also rcov 0.8.0.2 and

Hello all, I have two test beds running the exact same version of asterisk 1.2.7.1, latest of zaptel, libpri, etc.. Test bed #1 (Solaris 9,sparc ultra 5): This one is closer to a "production" machine, in that it is connected to a sip provider thru an iax2 connection and have an incoming DID configured. I can send and receive calls. Test bed #2 (Slackware Linux 10.2, AMD XP chip):

I am looking to issue a 'System cmd' that will echo the results into a new variable -- can anyone tell me the correct way to do this? example: at the terminal I am currently able to issue this command: echo | date > date.txt I end up with a file called date.txt with the contents: Mon Aug 29 18:32:45 EDT 2005 I would like to issue a command in my dialplan that puts the date

For some reason I can''t figure out why I am getting a routing error when I add Paperclip support. I have another test app when similar code is working. I created a new test app first without Paperclip support and it was doing the CRUD operation fine. However when I added Paperclip support I get the following error? Routing Error user_url failed to generate from

Hi, We are running asterisk 1.2.16 and need to connect two channels which are already established. We are currently using app_meetme to achieve that, but we are sometimes unhappy, as app_meetme provides functionality that produces load that we do not need in our two party conferences. I figured out that there is an alternative called app_changrab.

I have a "support call" AGI script that has been working flawlessly for a couple of years now. It dumps the customer into a MeetMe conference room, then dials a bunch of support engineers, and connects anyone who accepts the call into the conference room. The conference room is recorded. After the support call is over, the recording is emailed to a list for quality control, etc. It

Hello All, I posted a bug on the 14th of this month, and haven't heard anything back. However, I've since discovered that the problem is not in chan_iax.c as I originally thought, it's actually app_mixmonitor.c. Basically when I use 1.4.26.2 with an ilbc codec between two asterisk servers trunked via IAX, with mixmonitor Asterisk crashes on me. Here's a link to the post: