similar to: Channel failover

Here in Sweden, it's supposed to be springtime. A wonderful time of the year, with sunny skies and wonderful weather. Almost summer. Today, it's not. It's winter all over again with rain and only 3 degrees celsius outside. Better to stay inside and write a weekly Asterisk newsletter :-) This week's topics: ------------------- * Looking beyond Asterisk 1.0/1.1 - what's up? *

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

[Early Bird Ends Soon] ====================== AstriCon early-bird registration ends next Thursday, August 25th. Early-bird registration saves 20% ($110.00 USD) off the standard price of an AstriCon "All Access" Pass. This pass includes the pre-conference events (either the Meet Asterisk! seminar or the Asterisk Developers meeting), the tutorial day, both conference days, the exhibit

I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not

Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. Thanks, Gavin.

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

Hi all, I'm currently evaluating ideas for a phone system for our new office and I have a copy of Asterisk along with a couple of H.323 phones setup in the lab here. I'm very impressed - thanks for the great software! (I was especially impressed that the IAX connection to Digium worked out of the box like that - someone at Digium no doubt has a voicemail from me this morning!) ;-) What

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

Peter - 40 phones and only 3 PSTN trunks?. I would recommend at least 2 BRIs for this. If you have ISDN you can also get DDI to the extensions.I would strongly recommend abandoning the analogue PSTN lines and using ISDN. The setup pain you will go through will be significantly less, combined with better audio, more features (like DDI numbers!) and much more robust connections. You should look

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

>>>>> "MM" == Martin Maechler <maechler at stat.math.ethz.ch> >>>>> on Tue, 18 May 2010 12:37:21 +0200 writes: >>>>> "GaGr" == Gabor Grothendieck <ggrothendieck at gmail.com> >>>>> on Mon, 17 May 2010 09:45:00 -0400 writes: GaGr> BIC seems like something that would logically go into stats

I made myself look a little silly recently in a talk regarding asteriskdocs.org. I didn?t realize the 4th ed. of the Definitive Guide was apparently actually out (http://shop.oreilly.com/product/0636920025894.do), because I went by asteriskdocs.org?s claim that it was being worked on in OFPS (now retired, apparently.) Is the 4th ed. available to read online like the 3rd ed. was? Is someone on

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

Hi All I've done some reading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have

Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific. Don't (plain old) PBX' face the same issue if they use analogue interfaces? If there are analogue PBX' at all, how do they solve the problem? Yuan Liu

Hi, I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and wondered if anyone is able to offer any advice. In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk box. e.g: HiCom user dials access code and can call Asterisk extension or establish SIP call over Internet. Likewise, I'd like Asterisk to be able to present a call to the Hicom, either

Hi everyone, I want to get multiple DDI's and hunting across those DDI's in case one of the lines is busy using analog phone lines. The system is for a large house so I want 3 x PSTN lines. 3 x DDI's and the ability for those DDI's to be presented across all three PSTN lines. BT say you can't have more than one DDI number associated with a PSTN line and that you can't

Hello all, Can someone recommend what hardware to use for a 1000 analogue line capacity asterisk PABX? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/2bcd322f/attachment.html>