similar to: Inadyn error

Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 22, 2006 8:26 PM To:

Received this on the console -- IAX2/76.21.238.129:4569-4986 requested special control 20, passing it to SIP/magicjack-08225a58 Did a Google search, but reached a dead end Can anyone explain? Something need to be changed in my configuration? The call completed satisfactorily. Inbound IAX trunk - outbound to SIP provider magicjack ( no dongle ) Asterisk 1.4.26 TIA John Novack -- Dog is my

On Wed, Jul 29, 2015 at 10:16 AM, John Novack <jnovack at stromberg-carlson.org > wrote: > > > Murthy Gandikota wrote: > > > > ------------------------------ > To: asterisk-users at lists.digium.com > From: webaccounts173 at jgoettgens.de > Date: Wed, 29 Jul 2015 16:11:31 +0200 > Subject: Re: [asterisk-users] Windows Asterisk Help > > > >

If you are really wanting to build something on Raspberry Pi or similar ARM platform, you could also take a look at Elastix for ARM. http://www.elastix.com/en/downloads/ Elastix is a fully integrated platform, and includes the majority of necessary components in one installation. The new Raspberry Pi 2 platform may be perfect for your needs in this respect, although based on your load, the B+

Are other posters getting these annoying messages? Perhaps "serverhallen.com" needs to be removed ?? Posting this will generate yet another series of messages from their postmaster PITA John Novack -------- Original Message -------- Subject: Delivery Delayed: Re: [asterisk-users] Ring back tone with asterisk Date: Tue, 16 Nov 2010 18:02:16 +0100 From: <postmaster at

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

James Cass wrote: > I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for $30, and it works great for a home install. Very low power draw as well. > > James Cass <http://goog_987864563> > jcass78 at gmail.com <mailto:jcass78 at gmail.com> The JS-200 runs a very old ( 1.4 ) version of Asterisk I have set up more than 30 nodes using the HP thin

Hi, our Asterisk is connected to an E1 port. So we are using the DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for overlap digits for in-calls? I found the option "overlapdial=yes" but I did not try yet. Is that "my" option? Is there any option for setting an timeout? Thorsten

Hi C F no asterisk and sip device are not behind same router. actually both are in different countries. how ever when caller and callee are behind same routers voice is just fine (both ways) and i can see re-INVITEs too. but when someone calls from another router then this issue arises. caller can hear the called party but called party can not hear caller. and there are no re-invites issued

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms <http://voip.ms/> VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start

The mailing list will not receive emails from the forums. What I was referring to is that Discourse does provide the ability to receive emails for posts or threads you're interested in, and you are able to respond over email to them as well. On Mon, Dec 4, 2023 at 8:38 AM John Novack <jhnovack at stromberg-carlson.org> wrote: > > > Frank Vanoni wrote: > > On Mon,

Same about me - need to re-enable membership all the time. Annoying (( ??, 12 ???. 2017 ?. ? 15:59, John Novack <jnovack at comcast.net>: > Not just gmail > Happening as well with Comcast.net > > My Comcast address is set to forward to another domain, as Comcast seems > to now block sending mail with a non Comcast "from" address. they turned > that on a couple

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

Frank Vanoni wrote: > On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > >> To that end, we’ve decided to discontinue the mailing lists effective >> February 1st, 2024. > That's a very sad news! :-( > Agree. Yet another giant step backward. Interesting that they will continue to send e-mails when postings to the (UGH) forum happen though. John Novack -- Dog

I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

Just got a call from a company in Warren, MI . They recently had an Asterisk system put in by a vendor, and are having issues which need analysis and correction. They have a tremendous sense of urgency. They have about (40) users, and need DID's assigned to extensions and are having some echo issues at the site. If anyone is in the Warren, MI area, and is interested in some cavalry work,

I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.... Daniel Kocher

I'm wondering if the page application is broken in 1.2.5 The following: exten => 2001,1,Page(SIP/3254105) does strange stuff. The caller's phone immediately drops into the call, while the callee's phone is still ringing. I'd think it was a SIP messaging issue, except that the Dial() command is working fine, which makes me wonder if it's a bug in the Page appplication.