search for: jnovack

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

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

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 years ago with no notice...

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

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

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

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

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] Windo...

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:

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

...his 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 serverhallen.com> To: <jnovack at stromberg-carlson.org> *Delivery is delayed to these recipients or distribution lists:* asteriskusers at serverhallen.com <mailto:asteriskusers at serverhallen.com> Subject: Re: [asterisk-users] Ring back tone with asterisk This message has not yet been delivered. Microsoft Exchan...

...l, which makes sense if it's a SPA3k new dialplan feature. Cheers, Gonzalo. ------------------------------ Message: 3 Date: Sat, 22 Apr 2006 19:42:28 +0000 From: broadbandvoice@comcast.net Subject: Re: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3 To: jnovack@stromberg-carlson.org, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <042220061942.1215.444A87240001E385000004BF220073407608010B020E9B02@comcast.net> Content-Type: text/plain; charset="us-ascii" Thanks for the respon...

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

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,

Does anyone have an (overhead) paging system that they like that works with SIP? We've got a client with an old paging system that (supposedly) just takes an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn't auto-answer the call, so paging never happens. [cid:image001.png at 01D4DFF6.9C1F1AA0] Michael J. Munger, dCAP, MCPS, MCNPS, MBSS Microsoft Certified

Hi All... My Asterisk system uses VoIP and also 2 POTS lines from Cox Communications. Recently, the dial tone presentation from Cox seems to have slowed, so it can take as long as 3 seconds to get a dial tone. The problem I am having is that Asterisk does not seem to wait for the dial tone when dialing out. I'm using zaptel T400 cards. Is there any way to configure it such that I can

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

...may be perfect for your needs in this respect, although based on your load, the B+ board may be more available at this time, and slightly cheaper. The Pi 2 is about double the core processing speed. YMMV Thanks, *Glenn Geller* *VDOTel* On Fri, Mar 6, 2015 at 12:34 PM, John Novack SCII < jnovack at stromberg-carlson.org> wrote: > Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux > ( instructions at AstLinux.org ) Move your Digium card and your confs , fix > up any differences from your older version of Asterisk to the fairly > current version 11 cur...

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

Hello, I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog POTS lines coming into my Asterisk server from the phone company. Internally, I have about 180 SIP clients defined in sip.conf. What appears to be happening is that if existing calls are consuming all 8 external lines and a new SIP client attempts to make a call, an existing call gets dropped. The asterisk