similar to: ring (hunt?) group

Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming

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

Hello Everyone, I need to setup a dialplan where if a incoming call is rec'd to a number, Asterisk needs to dial several SIP extensions at the same time. The SIP extensions are for Cisco 7960s and each have multiple line appearnces. For example, exten => 9043442342,1,DIAL(SIP/102&SIP/103&SIP/104&SIP/105,,20) exten => 9043442342,1,Voicemail(u102) The issue I have is

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

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: 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

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

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 =>

Hello everyone, I'm sure someone had an experience arranging hunt-group setup for incoming calls on T1 PRI channels of Digium TE110P card. For instance, I have main DID channel associated with number (555) 222 0001. And I have whole bunch of other DID channels on same T1 card like (555) 222 0090, (555) 222 0091, (555) 222 0093. My goal is when a call comes to the main number which is

I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. -------------- next part --------------

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

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

Hello all, Quick description of scenario: Would like to be able to plug in an analog line to a Digium Wildcard X100P FXO card in an * server. Whenever that line receives a call, we would like it to automatically ring a hunt group of VoIP soft phones that are connected to the * server. Is this possible, and if it is, what's the easiest way to go about it? My thanks in advance. Gordon

Hi, Could someone please give me some advice on how to get * to perform the following :- Answer incoming call (from IAX Trunk) Play a prompt to the caller like "Thanks for calling please hold while you call is transferred to first available operator" The caller hears MusicOnHold Asterisk begins 'hunting' for someone to take the call by dialing through a list of

Come join the new Hunt Bugs Club. The web site is http://huntbugs.com/. The forum is on Google Groups at http://groups.google.com/group/hunt-bugs-club. This group will meet on the 3rd Wednesday of the month at Clockwork (http://www.clockwork.net). The next meeting is on Wednesday May 16th. The lightning talks will include: Bookmarklets: Andrew Leaf How to Write a Good Bug: Chris Countryman

Thanks ... I forgot to say I tried it with priorityjumping=yes in the [globals] section of extensions.conf still no go... Gerald, I'll try your suggestion, and try to figure out the result code tests :-) Thanks, Rich > -----Original Message----- > From: Gerald A [mailto:geraldablists at gmail.com] > Sent: Tuesday, October 16, 2007 23:59 > To: rich at isphone.net > Subject:

I was wondering if this is possible. I have a situation where I am connecting to a third party voicemail system from asterisk. I know this does not make since to everyone, but it has to be this way. Basically - I have an application that runs on the Asterisk system and when an employee calls into this system, they have an option to check there voicemail. This is where it needs to go over to

Hi I've been looking for 10 minutes and cant find dialparties.agi Can anyone tell me what folder this is located in as I'm going crazy. (if it makes a difference I use asterisk@home and am replacing the AMP dialparties.agi file) Super big TIA, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: