similar to: DAHDI / dial in / overlap digits / timeout

Hi, I'm trying to make a attended transfer through AMI. I though i could use Atxfer, and it seems ok, but nothing happens. And I can't find any how-to or description on how to do this. What more do I have to do to make this work? In Asterisk Call Manager: Action: Atxfer Channel: SIP/36-xxxxxx Exten: 33 Priority: 1 Context: Phone Response: Success Message: Atxfer successfully queued

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

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

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'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'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.

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 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

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one

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

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

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 > > > >

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

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

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

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

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