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On Sat, 2007-07-07 at 08:39 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Fri, 06 Jul 2007 12:02:53 -0600 > From: Stephen Bosch <posting at vodacomm.ca> > Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <468E83CD.1070703 at vodacomm.ca> > Content-Type: text/plain; charset=ISO-...

Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the

Hi: Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox server isn't seeing the mainboard's APIC. -Stephen-

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

Hi, folks: I need to be able to have a single DID ring multiple remote (IP and PSTN) extensions, and then pass the call to whichever picks up first. I'm sure this is old hat -- lots of providers offer it. I see that Trixbox will do it, but it's not clear how it's doing it. They use different terminology -- a "ring group" and "hunt strategy" How can it be done

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I

Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere

Hi All, I would appreciate a lot if you could help me. I have installed Asterisk 1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have also installed 1 FXO port card: Digium TDM400P. After loading zaptel driver I could see my digium card's led glow green. Tested with zttool that its in OK state. I have configured fxsks=4 in zaptel.conf(channel 4 cause FXO module is on port 4). I

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

...caller can hear this and still have the 2 or 3 second window to hang up and get his $$ back. This emulates PSTN behavior as close as possible but you have to train your users to hit the DnD button when they walk away from the phonw. -----Original Message----- From: Stephen Bosch [mailto:posting@vodacomm.ca] Sent: Monday, June 12, 2006 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line? Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in...

Hi: This should be easy. I'm running 1.2.15. When a caller calls someone's voice mail, it goes straight to a beep, even though there is an unavail.wav file in that user's voice mail directory. Here is the relevant part of extensions.conf: [internal] exten => 2211,1,Dial(SIP/211,10) exten => 2211,2,VoiceMail(u211@default) exten => 2211,3,Hangup Here is the relevant part of

Hi, Anybody lucky with this config inside an Asterisk server for dealing with FAX ? FXO_LINE ----> ASTERISK 1.4.2 ---> IAXMODEM ------> HYLAFAX TDM400P ZAPTEL 4.3.1 1 FXO port 1.41 ------------------------------------------------------------------------ I know Fax is not officially supported on TDM400P cards but I did not expect not being able of sending one

...number plan zone prefix THORCOM 9.. gw-priority 10 PABX ! emergency calls 911/999 gw-type-prefix 1#* default-technology no shutdown ! ... so I think this should route the calls between the gateways...? Mike ----- Original Message ----- From: "Stephen Bosch" <posting@vodacomm.ca> To: "Michael J. Tubby G8TIC" <mike.tubby@thorcom.co.uk> Sent: Monday, February 05, 2007 7:51 PM Subject: Re: [asterisk-users] Help sought: Asterisk H.323, Cisco IOS Gatekeeper(s) intra-zone call routing and TETRA > Michael J. Tubby G8TIC wrote: >> I can see how th...

Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some

Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen-

Hi guys, I know it's a little off topic but......Wondering if you can help. My wife has been asked to find a writer to produce a story on "The dramatic ramifications of IPV6 on commercial businesses and how it will change the product designs for ordinary household/commercial use in a 5-10 year time frame" So her company hired someone who should have been able to deliver the

I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ame ike If the taps are set to less than 16, there

We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's something specific that I need to do to make it work, since many PBXs can do this. The switch is a

...les available to the community, and take up donations for the studio and processing time. Talk to you soon. I'm going to cc the asterisk-users list for this, so that the community can benefit from the information. SKM ->-----Original Message----- ->From: Stephen Bosch [mailto:posting@vodacomm.ca] ->Sent: Thursday, September 29, 2005 12:18 AM ->To: Sherwood McGowan -> ->Hi, Sherwood: -> ->If you'll forgive me, I'd like to e-mail you directly with a ->few comments and questions. -> ->Sherwood McGowan wrote: ->> I have to barge in here... ->&gt...

Hi, folks: I remain intrigued by the gap in BRI implementation between North America and Europe, and I wanted to get feedback from the list members on the matter. I'm seriously considering making the leap in our office. In Europe, the idea that an office that does not have enough lines to justify PRI would use analog lines is perceived as technologically backwards, and yet that's what