similar to: Asterisk/app_rpt and bandwidth

I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

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 As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code.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

Dear All, I am not sure if this is the right place to ask my question but I can't find a newsgroup or support for this app_RPT concept so I hope if some one in this community who have tried it out could help me out. I studied this application requirments and saw the hardware needed they describe a radio quad which uses RJ 45 but I can't see where the RJ goes in order to be able to

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

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

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

"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.

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 :)

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

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

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

I have a server with 8 Xorcom Astribank, they're connected to the server via a USB hub, my problem is: when a reboot the server the ID of the Channel Bank changes, cousing a big mess on my server. Anybody know how can I solve this ? Best regards, -- Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre:

I'm trying to install the SPANDSP software into Asterisk to support incoming (mainly) Fax. I'm following the info in http://www.voip-info.org/wiki-Asterisk+Fax. I downloaded and installed the spandsp software from ftp://ftp.opencall.org/pub/spandsp/ and followed the directions in several documents listed on the on the Tiki page. I get down to patch < Makefile.patch that fails with

Hi there, I was looking for a GSM gateway, but I didn't find any prices... Anybody knows how much one of these costs? And, is it possible to use an amateur radio with asterisk? Thanks, Marconi.

So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230908/dee530c8/attachment.html>

I'm trying to produce some enhancements to one of the applications, and am trying to use ast_request("zap", format, "pseudo") to create a new channel on /dev/zap/pseudo, which I can then bind to a zaptel conference and play a stream to it. I've been using as inspiration the Radio Repeater app, app_rpt.c, which uses this technique to play idents and announcements.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Have anyone tried using this? I've looked at app_rpt, and that's a nice project, but have anyone tried using Asterisk for radio services using a Mux e.g.? I was thinking of using an E&M Mux (or channel bank i think) with TX/RX/BUSY/PTT functionality. Or even tried to decode any signalling commonly used with radio communication?

Daniel Nystrom wrote > It seems like the Radio discussions is closing in on something I was > interested in. > How about controlling 30 2-way radios via E1 and 30-channel "Mux" > (channel bank?) with E&M signalling? > I think the Mux uses CAS and each channel has Audio out, PTT, Audio > IN, Busy. 6-wire connection i guess? > That should be a really nice setup