similar to: Nokia PoC PTT Asterisk

Hello all, Can asterisk be used to provide PTT feature? Anyone using or tested it? Thanks: -AEE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100911/49983cdc/attachment.htm

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>

>Hi Jerry > >As others have eluded to, the 'PTT' feature can mean different things to different >people depending on their background.> > >Is it fair to say that you're looking for a one-touch button which initiates a call to >the other end and causes the other end to automatically answer in speakerphone >mode? >If that would foot the bill then have a

Hi, We recently created a Ruby on Rails POC, which landed us with a client. More about this initiative at http://www.articlesbase.com/information-technology-articles/cpal-a-ruby-on-rails-poc-1316554.html Thanks, Miti

-----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

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

# HG changeset patch # User Ian Campbell <ian.campbell@citrix.com> # Date 1294742832 0 # Node ID 03997417771be2aa122d7dac31256399f854c5b1 # Parent b279ec057736ead1d7061f8ec8a29a8676dd56b4 PoC: libxc+ocaml: add interfaces to send triggers to HVM guests * ACPI power button event * ACPI sleep button event Required by xapi. Taken from xen-api-libs.hg/xc and adjusted for upstream libxc+ocaml.

Run: # top # vmstat 1 # iostat -x /dev/emcpowerb 1 The latter two you can save to a file. For top, just monitor cpu usage and see if any process is hogging all of it. Colin Laird wrote: > and the fstab settings: > > # This file is edited by fstab-sync - see 'man fstab-sync' for details > /dev/VolGroup00/LogVol01 / ext3 > defaults 1 1 >

Hello all, I am trying to implement a POC wherein I can post a message + image and get the same working via AJAX to display them both for each new entry without the page reloading. I am using a config of Ruby 1.8.7 and Rails 2.0.2 for project specific purposes.. I am also using a pretty old paperclip commit from github for this....I had to suit my proj config requirements mentioned previously..

So, incoming calls on zap work just as I expect them - an intro is played, the caller hits 1 for sale 2 for support or dials an extension. I'm using the privacy option for all extensions. When calls come in from zap, they caller is played the priv-recordintro recording, they say their name, and everything happens normally from there on out. However, when the call comes in from sip and

Hi, I have successfully configured an IAX trunk between 2 asterisks, calls can go through both ways without any problems, NAT in the middle of course (iptables) Now , leave them for a while , and make a call from the external server , it doesn't go through, Dial from the internal one, everything works fine again.. Now , it is clearly a problem in the NAT engine,

> Date: Mon, 2 Feb 2009 12:56:15 +0100 > From: friedrich.leisch at stat.uni-muenchen.de > Subject: Re: [R] Problems in Recommending R > To: thomas.petzoldt at tu-dresden.de > Cc: "r-help at r-project.org" <r-help at r-project.org>, > useR-2009 at r-project.org, paul at stat.auckland.ac.nz > Message-ID: <18822.57183.637787.426445 at

"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,

Hi, In case you are interested in testing the Doveadm HTTP API in a bit more simplified fashion I have written a client for it in Python. Feedback is always appreciated (pull requests more appreciated). https://github.com/hnsk/doveadm-http-cli <https://github.com/hnsk/doveadm-http-cli> -Hannu

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.

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

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:

I am starting wine with the command: wine explorer /desktop=Name,640x480 PPTVIEW.EXE myppt.ptt and I get the following error: [silentm@geisjdell PowerPoint Viewer]$ fixme:actctx:QueryActCtxW 80000010 0x3018b4d0 (nil) 1 0x34fb60 8 (nil) X Error of failed request: BadWindow (invalid Window parameter) Major opcode of failed request: 1 (X_CreateWindow) Resource id in failed request:

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid