similar to: Need help with getting EXTEN from pstn hunt group

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

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

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

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

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

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.

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 --------------

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

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

I am thinking of having a pots line with multiple numbers on it, and having Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring another desk if the person called xxx-xxx-xxx2, etc. Can Asterisk do this? -- Jeff Ramsey MIS Administrator Tubafor Mill, Inc.

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

Hai, I know, im not the best in english, but i've updated the time sync page on the wiki. If someone can check it and update where needed. I've added this part. https://wiki.samba.org/index.php/Time_Synchronisation#Requirements_2 And https://wiki.samba.org/index.php/Time_Synchronisation#Set_up_time_with_systemd-timesyncd_on_a_Unix_Domain_Member Greetz, Louis

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 =>

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 was wondering two things: I am running Asterisk Current Stable. Is there a way in Asterisk or using other equipment or through the TelCO that I can make a channel bank such that say I have 3 Phone number either with the use of FXO Cards or by using 3 connections via IAX to a service that could then provide PSTN connectivity where when Line 1 is called the call is then switched and held on

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

Hi all, Newbie quest here. I have 3 PSTN lines (2430-2432) setup by the CLEC in a hunt group coming in to a TDM04B and 3 Grandstream gxp-2000's (say a,b and c) and Asteriskathome installed. I want all phones (a,b,c) to be able to take calls from the hunt group. Does this mean I must make 9 SIP extensions on * ? 3 for each IP phone? and group them in trtiplets (ring groups?)

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

Is there a way to setup an IAX bat phone (immediate=yes) or is this a privilege only reserved for ZAP channels? -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/6ff27919/attachment.htm