search for: novack

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.

...ginal poster was asking about which OS is best. I think he was asking which PBX manager people are using. Ex, PBX in a Flash, Elastix, FreePBX, blah, blah, blah. Thanks; John From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Novack Sent: Wednesday, December 23, 2015 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Asterisk Platform Are you trying to start a religious argument? CentOS /RedHat appear to be the most trouble free when compiling from source JMO John Novac...

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

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

...ular cadence? Eg; will there be an AMI for 7.3.1611, for example? I realize that we could do a yum update, but we'd prefer to start off new hosts on the latest version without needing to do a yum update and then save our own image. Any insight is appreciated and thank you! -- Thank you, Wes Novack -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos-virt/attachments/20170124/8e053afb/attachment-0002.html>

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! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/4206395a/attachment.htm

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

...gt; >> 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 is my Co-Pilot

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

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

...o dial before dial tone is returned from the CO This was determined by monitoring the CO line with a buttset Since this has been talked about on the list before, I have to wonder if is version dependent, or if it works at all. Anyone KNOW that this works? And with what version of Asterisk? John Novack John Novack wrote: > >>>>> Or, in the example below, wait before dialing? >>>>> >>>>> exten => s,1,Dial(ZAP/g1/${ARG1},360) ; ARG1 is the number to be >>>>> dialed >>>> >>>> >>>> >>>&g...

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

...| 0      | | Ssl_finished_connects    | 0      | | Threads_connected        | 3      | +--------------------------+--------+ 7 rows in set (0.00 sec) I stick to 1.8 because it just works. I had some issues with version 11 and 13 in the past. Regards Jonas. Op 04-10-18 om 17:49 schreef John Novack: > Woefully out of date. > You really need to put your efforts into at least a modest upgrade > I use version 13 with MySql queries built into the dialplan on CentOs > 6 and have NO such issues, either performance or any restart or > reboot. It simply works > > I never used e...

Hi, I’m currently using 32 CENTOS 5 but it’s now unsupported. I only have a 32 bit processor and CENTOS no longer supports 32 bit so I need to move on. I’ve installed the current version of 32 bit Fedora and I can’t get the latest Dahdi to build. Even tried downloading the early release DAHDI from github with no luck. Any recommendation for which 32 bit LINUX to use going forward? Or

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,

What is the best asterisk platform to use? What are you guys using? I am looking for something to host either in our data center or at the customer prem where I have the control over the unit and not through a contractor. I dont mind paying a license fee for a front end interface but still would rather not have to pay. Thanks, --Eric -------------- next part -------------- An HTML attachment

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