similar to: Hunt group with Accept/Reject Option

Hello anyone who can help I have two Asterisk boxes with identical hardware (Dev & Production). I recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head. The hardware is an Intel CA810e, onboard everything with a PIII processor. The config is pure VOIP using IAX2 & ilBC with Virbiage Firefly soft clients. I also use Ztdummy which seems to be working ok - no error

Hi I have a TDM card with 4 lines on a hunt group coming in. I can answer the phones with exten => s,1,Answer() exten => s,n,Dial(Zap....&Zap....) ... The problem is I don't know how to find out what extension was originally dialed. And, trying to match on the extension always fails. E.g. exten => 1234567,1,Answer() # never gets here I thought I could get the extension on

Hi All, I'm trying to configure a Digium T100P to talk to a legacy voicemail system. I have the signaling specs verbatim from the original manufacturer documentation as follows: [T1 Signaling] Service Type: T1,D4 format, AMI(Super Fram) Signaling: Four wire, terminated, E&M (Robbed bit) Start Protocol: Wink start; 250msec duration Dial Tone: Enabled Digits: DTMF, 4-digits DTMF: 50msec

Does anyone out there know if it is possible to discover the dialed number when a line in an analog hunt group rings? I can't get a straight answer from our IT folks. We have a 5ess switch delivering 4 analog lines which are in a simple hunt group servicing our lab. I would like to have a different call attendant based on which number is dialed so that I can route the calls to the

Hi, I've set up a Gizmo Project account for access on my Nokia E61 because they work through NAT. Trouble is If I include my gizmo account in an asterisk hunt group and I'm not connected (phone is off / outside wireless coverage) the gizmo project always answers. Either the call goes to voice mail or if I turn voicemail off the call gets answered by a recording saying I'm not

I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments suggestions? I have not had much luck with the outgoinglimit=1, incominglimit=1 stuff that I would need to get busy extinctions to work right, which is why I'm asking on the list.

Hi, all. I've got an Asterisk box installed that I'd really like to leverage -- and installing a GUI for hunt groups would be awesome. So long as I can have a trial copy, I could even pay money. It would have to be able to make use of both SIP and ZAP extensions. Suggestions? (Note: I wouldn't much care about the GUI, myself, but my boss is all over one.) Thanks! -Ken -- This

Hi, Is doing this as simple as just creating a queue in queue.conf? I have the following setup: 1. Single hunting 1-800 number mapped to multiple numbers in a hunt group by the telco. 2. All calls land up at Asterisk over SIP. Now I need: 3. A hunt group of operator extensions mapped to the incoming lines. 4. Music-on-hold if all the operator extensions are busy. Queues seem to handle this

Hi , where can i find examples of dial plans using ring/hunt group feature? thanks

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

You should be able to get ISDN though. I think BT provide this anywhere. Otherwise your solution will be a Primary rate (ISDN30) with 8 channels. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matt Sent: 07 June 2004 10:19 To: asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] Multiple DDI &

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 All, I defined 3 trunks for a client: [trunk-100] ... [trunk-101] ... [trunk-102] ... How can i do a trunk hunting ability like this: [dial-out] exten => _1.,1,Dial(SIP/${EXTEN}@trunk-100) if busy try: exten => _1.,1,Dial(SIP/${EXTEN}@trunk-102) if busy try: exten => _1.,1,Dial(SIP/${EXTEN}@trunk-103) How can i make it dynamic,coz different clients will have different number of

I see "Blue's Clues PreSchool" on the AppDB, with the note that it installs and runs OK. I have "Blue's Treasure Hunt" by InfoGames (<www.us.infogames.com). When I mount the CD, it shows the "Preparing the InstallShield wizard" box, which quickly goes to 100%. However, I then immediately get an almost empty dialog box with the Error icon (no text).

I havent seen a post here about this so here is mine..... Wine is nearing its 1.0 release, and we need your help to make sure it's a good one! Wine has been under heavy development in recent months, and some applications that used to work well no longer do. But we don't know which ones! Please help us find them, so we can fix them. Here's how:

I have a question about how Asterisk Parses the Dial Plan. To create a hunt-group which would be the appropriate dial plan: [CompanyABC] exten => 7228888,1,Dial(SIP/8017228888,60,r) exten => 7228888,102,Dial(SIP/8014361234,60,r) exten => 7228888,103,Dial(SIP/8014362345,60,r) exten => 7228888,104,Dial(SIP/8014363456,60,r) exten => 7228888,105,Dial(SIP/8014364567,60,r) exten

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go

On 31 March 2016 at 21:41, Mehdi Amini via llvm-dev <llvm-dev at lists.llvm.org> wrote: > TLDR: I totally support considering compile time regression as bug. Me too. I also agree that reverting fresh and reapplying is *much* easier than trying to revert late. But I'd like to avoid dubious metrics. > The closest I could find would be what Chandler wrote in: >

*My apologies if this message is posted 3 times, I was trying to sent it to the list once before I am a list-member, the second time before I was approved. Can anyone point me to some resources on using hunting lines with Asterisk? Sales support of my telco have no idea what I am trying to do. They asked what pbx system I am using, I was like Aster... never mind =) I am trying to setup

Asterisk 1.4.2 I have spent much of today trying to make a DID (from SIP GW) ring to 4 extensions in a hunt (roll-over) group. Results from searching docs and forums seem to indicate it is doable and so trivial no one includes an actual example. I can make all 4 exts ring at once with the like of exten => _1655,1,Ringing() exten =>