similar to: Unable to transfer external calls to MeetMeconference (re-post)

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP

Hi All; Is there any limitation on the number of users for MeetMe Conference? In other words, how many parties can join to the room and become a member of the room? Is there any limitation? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

I am curious to know if anyone has OPIE, S/Key, or any similar open source One-Time Password or Dual Authentication system working on Centos 4. Sincerely, Trevor Hammonds SkyHost / SkyNet Information Systems

Does anyone know of a AGI script that takes advantage of the weather sound files that's included with the extra sound files available from www.loligo.com/asterisk/sounds/ <http://www.loligo.com/asterisk/sounds/> ? Thank, Jeramie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I want to do a scripted 'restart when convenient' on a daily basis. This used to work, but since i've upgraded to Asterisk 11.7 it seems it's never convenient to restart the server. My question: how can i tell *why* it's not convenient to restart the server? It used to be some colleague left the receiver OffHook or something like that, but even when i'm fairly

Am 13.06.2015 um 21:22 schrieb jd1008: > > > On 06/12/2015 01:06 PM, Jonathan Billings wrote: >> On Sat, Jun 13, 2015 at 01:04:03PM -0600, jd1008 wrote: >>> My computer clock is only seconds ahead of my phone, >>> which is timed to the cellular provider's ntp server. >> Your clock is an entire day ahead of your cell phone, unless you're >>

On 06/12/2015 01:06 PM, Jonathan Billings wrote: > On Sat, Jun 13, 2015 at 01:04:03PM -0600, jd1008 wrote: >> My computer clock is only seconds ahead of my phone, >> which is timed to the cellular provider's ntp server. > Your clock is an entire day ahead of your cell phone, unless you're > emailing us from the future. > > > If you are emailing from the

On 06/12/2015 01:24 PM, Alexander Dalloz wrote: > Am 13.06.2015 um 21:22 schrieb jd1008: >> >> >> On 06/12/2015 01:06 PM, Jonathan Billings wrote: >>> On Sat, Jun 13, 2015 at 01:04:03PM -0600, jd1008 wrote: >>>> My computer clock is only seconds ahead of my phone, >>>> which is timed to the cellular provider's ntp server. >>> Your

I have just the thing in PHP. Drop me a personal e-mail and I'll whiz it over. Andrew Thomas Technical Services Manager andy at datavox.co.uk DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Trevor Hammonds

Hi I have a problem with the queues on Asterisk. The setup is Asterisk@Home v1.0 with Asterisk 1.0.7. I have 1 queue (4500) set up, with leastrecent strategy. There are no agents configured in this queue. Agents log in by dialing 4500* on their phones. All incoming calls are sent to the queue. Calls wait 120 seconds in the queue, and are then sent to voicemail extension 310. My problem is

I set up certain routes in my ASTCC application to be free of charge. When a user attempts to dial one of these numbers, the announcement plays the prompts "This call will cost", "nothing", and then terminates the script, dropping the call, leaving the card locked in the database as being in use. Any ideas? Sincerely, Trevor Hammonds -------------- next

I have a couple "headless" servers with some spare bandwidth. What is the best way to seed multiple CentOS ISO torrents? Thanks for any assistance. Sincerely, Trevor Hammonds

I have an Avaya 4602 IP phone that was previously working with Asterisk. It was being used elsewhere for several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP

great info on the dna lounge site, but very discouraging. o, basically, we would not be able to do this unless we wanted to pay a ridiculous amount of money. we could keep it on the down low, but this kind of defeats the owner's purpose for the stream, as he wanted to advertise it so potential visitors could check it out before they come. is there no way to make this happen? (legally) jg

hello i'm new with icecast2 but allready addicted to this nice code. i installed icecast2 a few weeks ago an its running stable in my test environment. i'm thinking on a productiv switch for the next month. for our radio station i'm searching a php script, which is able to parse the XML data and shows up the last played tracks. has anyone of you allready did such a script and would

can't find help on this anywhere! using ice get error message from ice: [ ices]$ bin/ices -F ~uo/dilated.txt -P ***** Logfile opened Mounted on http://127.0.0.1:8000/ices Playing /export/mp3/My Music/Rawkus/Lyricist Lounge 2/Dilated Peoples - Right and Exact.mp3 Error during send: Libshout reported send error, disconnecting: Libshout socket error. Mounted on http://127.0.0.1:8000/ices Ices

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 DeveloperCube is a new project started by veterans of the web development industry. We are proud to announce that we are now the Official Shorewall Support Forum. We are an online community offering discussion geared towards web developers, designers, and administrators of all skill levels. There are topics ranging from how to market your website,

I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the

boah how i hate that "reply all" attitude leading to break "reply to list" and leads in off-list replies -------- Original-Nachricht -------- Betreff: Re: [Dovecot] Dovecot not honoring configuration settings (auth failure) Datum: Tue, 01 Apr 2014 00:02:42 +0200 Von: Reindl Harald <h.reindl at thelounge.net> Organisation: the lounge interactive design An: noloader at

I'm hoping this is the last issue I run into with bringing this new DC online. DNS replication is currently only working in one direction, from my old DC to my new DC. Items added or changed in the RSAT of my new DC don't ever make it over to the old DC. I have turned up samba logging on each side to 3, and you can see the logs below from the time I created a record on the new DC