similar to: Asterisk with app_RPT question

Hello to the list, I have a question about Asterisk/app_rpt: When a signal is *not* being received and or transmitted by the radio system attached to Asterisk/app_rpt via its interface, is the incoming and or outgoing data suppressed (silence suppression)? In other words! At idle but still connected to some other device, what data will be transferred? I currently don't have access to a

Greetings, Where can I get the chan_echolink channel driver from? I've seen reference to it, but have yet to find a place to download/compile it. It is part of the app_rpt.so module... I am told, but do not see the source with app_rpt. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help

Hello list, I am using the Microsoft ADUC (Active Directory Users & Computers) tool from the RSAT suite for creating and modifying my domain users. I am aware of the "copy" functionality which really is very nice to use. Unfortunately I am missing two important actions during the user-creation process which I try to describe: 1.) When I use the template feauture (by using the

This is just curiosity, but I'm wondering why the Morsecode app has remained part of the trunk for all of these years. Is there any practical use for this or is it just an homage to the ghosts of telecommunications past? Does anybody use the Morsecode app for anything interesting? I'm strangely fascinated by this core piece of Asterisk functionality. -Chris

Hello, I have a mixture of Polycom SP IP 500 and 300 phones. I have been reading through the administration manual to try and solve this problem, but I do not seem to be able to find the answers to my question. I figured I would ask here and see if anyone has some suggestions. The problem is kind of annoying. After dialing 4 digits, the phone seems to pause and miss the 5th digit, often

I switched my file server's hard disks from a p-2 350 to a p-3 deley 1.1 ghz. I ran kudzu and it detected the new nic and unconfigured the old one. I may have messed up the static ip assignment because hte system is not able to do anything on the network. How do i check the ipaddress asignment and also ensure the proper linksys LNE-100TX driver is loaded? thanks. William -- Computer

Good day. I want to migrate from cyrus-imap to dovecot but I can not find replacement of one feauture presented in the cyrus-imap. It is LMTP. All mail users are virtual and mail works like: username at real-domain.com -> anotherusername at fake-domain.local. I whould like to check "delivery is possable" in the exim's ACL. In the exim+cirus scheme I use Callout verification from

Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk??

Hy, I have a network with WIFI communication and VHF/ HF channels. I have integrated asterisk in the network using SIP, ZAP and IAX2 channels for WIFI communications, but I don't Know How I could integrate the VHF/ HF channels. I have heard speaking about app_rpt project, but I don't Know very much about this. Could I integrate VHF/ HF channels with this application? if the answer is

"app_rpt is an application which comes bundled with Asterisk, however, a later version may be available on our source repository. All you need to do is go to asterisk.org, download asterisk, configure the asterisk to build app_rpt by modifying the Makefile in the asterisk/apps directory, and then compile and install it. You can get the latest version of app_rpt.c along with the sound files,

Pete VK2YX writes > I've been involved with IRLP for about 5 years and am one of the original > install team. I've gone through the emmotions of allowing "other" networks > connect to IRLP and I know its caused some lots of headache. > > As far as a closed network goes, yes there is LOTS of passion to keep it HAM > only and I'd have to support that notion.

I'm tring to compile fax applications on Debian system. the spandsp library compiles ok, and when i try to patch the make file in apps directory as is said in the instructions it returns errors. I'm using cvs version of asterisk . -------------------------- voipgw:/usr/src/asterisk/apps# patch < Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of

Here''s an excellent blog post on Story Runner, which will be part of the next release and is undergoing active development in trunk: http://evang.eli.st/blog/2007/9/1/user-stories-with-rspec-s-story-runner

I'd like to flag a problem we are observing that looks like it is caused by the listescape-plugin.c. I realize that this plugin is not part of the standard Dovecot 1.0 distribution but I would appreciate any insights or suggestions for fixes that anyone is kind enough to offer. The problem concerns subfolder renames when used in conjunction with Maildir folders (I've not tested this

Any one know of software that allows 2-way radios as VoIP(SIP) clients, besides dingotel's usb & mic cable trick ? http://www.dingotel.com/2way/requirements2way.asp They might be ok if the SIP client was not hardcode to their own SIP proxy Has anyone tried any hacks to get the 2-way radio SIP client to regsiter to a * box. hmm chan_frsgmfrs anyone? using the usb/mic cable under linux :)

Hi all, This my first mail in the community. My name is Manolo. I'm new in Asterisk. My objective is to control some radio using Asterisk via web. In google I could see that the application app_rpt had this goal, but currently is not in the new Asterisk 1.8 or higher. Thinking in future, I thought the easer way to do that is to buy some RoIP unit and do the communication with

I've installed spandsp-0.0.1k on a RHv9 box with CVS-HEAD-09/19/04 and compiled the libraries just fine. Having a problem with patching the asterisk/apps Makefile however. The patch attempt results in: [root@phoenix apps]# patch <Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej [root@phoenix

You don't need to reinvent anything to tie radios to *. Ham systems like IRLP, Echolink, eqso etc all have fairly tight controls to keep from being abused (although with a little Linux knowledge, the IRLP package can easily be used to set up your own network using their protocol). Jim Dixon seems to have done the work to integrate radios with * already. See

Hi, is it possible to install ocfs2 in centos-5 without build a new kernel? Regards Joachim Backes <joachim.backes at rhrk.uni-kl.de> University of Kaiserslautern,Computer Center [RHRK], Systems and Operations, High Performance Computing, D-67653 Kaiserslautern, PO Box 3049, Germany -------------------------------------------------- Phone: +49-631-205-2438, FAX: +49-631-205-3056