similar to: [Invitation] VoIP Users Conference @ Fri Mar 7 12:00 - 13:00 ()

Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Tomorrow, Pika Technologies will be with us. Friday, March 7that 12:00 PM (Eastern US) 9AM PST, 5PM GMT *** Call (724) 444-7444 or SIP:123 at 66.212.134.192 *** After the call connects, enter the conf: 22622# and your_PIN# (or 1# if you

Please visit our family photo albums at http://www.photos.vanpeters.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080306/9c42cd2b/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/calendar Size: 1524 bytes Desc: not

------------------ New Meeting Time Proposed: 10 March 2010 12:00-13:00 (GMT+01:00) Brussels, Copenhagen, Madrid, Paris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080306/df759d11/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Ottawa, Canada ? February 15, 2006 - PIKA Technologies Inc. today announced that they have integrated PIKA?s high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability

Happy coming Spring, Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Friday, February 22 at 12:00 PM (Eastern US) 9AM PST, 5PM GMT * Call (724) 444-7444 or sip:123 at 66.212.134.192 After the call connects, enter the conf: 22622# and your_PIN# (or 1# if you have no PIN)

Leap year? Election year? Will your GoToIfTime() dialplan function properly on Feb 29th? Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Aastra has some interesting phones so we asked them to come talk about them. Friday, February 29 at 12:00 PM (Eastern US) 9AM PST, 5PM GMT * Call (724)

Greetings to all. I am not sure of this is a "user" question or a "business" so apologies in advance if it should be asked in the business list. A client of mine has a UK branch that is served by a provider that uses the Broadsoft solution. I want to create a sip trunk from a remote asterisk pbx to the client. The provider claims that Broadsoft has a strict

Don't miss the call this week, Friday at 12 Noon EST, 9 AM PST, 6PM Western Europe. About the conference (formerly Asterisk Users Conference, Asterisk is a registered trademark of Digium) * http://www.voipusersconference.org * There is a Flash player on the above page. If you hate Flash, you can download any recordings in mp3 format here: * http://food4wine.ning.com/conference * Two

Friday March 28th;: Astlinux How to be a part of the conference via PSTN or SIP : http://www.VoipUsersConference.org Astlinux is a custom Linux distribution that is centered around the goal of providing Asterisk. Astlinux is highly optimized for Asterisk both in commercial and embedded systems. They will be releasing a new version shortly and Darrick Hartman has graciously offered to be a

Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. Thanks in advance! -- -------------------------------------------- Paulo Garcia Pika

Hello! I'd like to use PriorityInheritance option, despite it is still experimental. (Why it could be experimental, when it looks quite simple feature, no?) But this option doesn't change the TOS field of outgoing UDP packet. I just do tcpdump on outgoing interface from vty1: $ sudo tcpdump -vni eth0 udp dst port 655 And from vty2 I do ping with setting of tos field to EF(0xb8). $

Hi All, At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael Robertson to join the discussion to filed questions about OpenSky and Gizmo5. I have been testing all of these Skype to X methods except SIP for Skype since I have no word from them. I can tell you that we've had good results with bith Skype for Asterisk and OpenSky. In fact, I am currently accepting calls to my

I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com "Realize the Value of Customer Contact!"TM This e-mail is intended

Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there "WARP" appliance. NOT really looking to migrate from 1.4.x to 1.6.x -------------- next part -------------- An HTML attachment was

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</font> <ul class="D"> <li class="D_off">Two Available PCI Express x8 Slots</li> <li class="D_on">Two Available PCI Express x8 Low Profile Slots</li> <li class="D_off">One Available 64-bit/100MHz PCI-X slot</li> </ul> The list has already answered what goes in what slot so I won't repeat that.

Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 2016-08-29 12:28, Eric Klein wrote: > Hi Hooman, > > What you probably want is a PRI PBX running Asterisk. > > You should either plan to build your own (with the cards you need) or get one of the low cost options: > > * Allo.com has their Mega PBX with 1 PPR port (http://allo.com/megapbx-line.html) > * Pika Tech has the Warp PBX with BRI

Hi, I would like to understand how those features (subject) work on fxs ports. Unfortunately I don't have a digium card with this kind of port, then any help will be appreciated. I tried to gather some information from google and this list history, but I still need some help. 3-way-call - As I could understand, when you are talking to A-person, you can press *flash*, call to B-person and

Hi, I'm currently having problems detecting FSK BT (UK) caller id in our API (Pika boards). I have a recording to test on but it is giving me checksum errors. I'm wondering if someone from UK using BT lines could send me a recording with the FSK signal so I can have more data to work on? If you have something, please send it to paulo.astuser at gmail.com. I appreciate all help about