similar to: Camp On configuration?

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses

Can the Manager Redirect command transfer a ringing SIP extension? I'm trying to implement a Camp On feature, and having failed to do it in Dial Plan logic, am trying to do it with manager logic. If an arbitrary Sip extension is ringing, I need the ability to pick up that extension from any other phone. What little docs there are on Manager commands shows Redirect takes these parameters:

From: "Nik Martin" <nmartin@radiancetech.com>> To: <asterisk-users@lists.digium.com>> Subject: RE: [Asterisk-Users] asterisk compile problem Date: Wed, 14 Jul 2004 09:22:38 -0500 Organization: Radiance Technologies, Inc. Reply-To: asterisk-users@lists.digium.com Fletcher Bonds wrote: >> Hello all >> >> As of 5pm PST today (7/13), I pulled

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hi all After the success of last year''s Puppet Camp and Puppet Camp Europe we''re going to be running Puppet Camp again in the United States. The current dates are October 7th and 8th (Thursday and Friday) and the location will be San Francisco - currently TBD but downtown San Francisco. So mark your diary! Like previous Camps there will be a small cost to cover food and venue

Hi all Due to the late pull out of one of our potential speakers we have a speaking spot open for Puppet Camp in SFO (http://www.puppetlabs.com/community/puppet-camp/puppet-camp-sf-2010-info/). So this is your big chance to make your mark on the Puppet community and tell us about the cool thing you do or the quirky Puppet implementation you have. We''re looking for 45 minute talks

Hi, Is there any way to pick up a remote phone/extension wich is ringing for a long time but no one is availabe to answer that call? OR the scenario is like, suppose my extension is 1000, and I am working at a colleagues desk. At this time my phone is ringing in my desk and I want to pick up the extension I am presently at and get the call. I have another question about the remote login. Is

Bar Camp Chennai-India is happening, get to know the latest in the Web 2.0scene in India, to register/help move here. http://barcamp.org/BarCampChennai -- Warm Regards Young J -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060314/a419d5b1/attachment.html

If I have an IAX client (Firefly softphone in this example), and the client is not registered at the moment because they are not connected to the network and someone dial that extension, they get the user's "I'm on the phone at the moment" message vs. the "I'm unavailable" message. Is this by design? Here's the extension in question's dialplan:

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

Hello, Reevik Technologies Pvt. Ltd. will be conducting a 5-day training camp on Ruby and Ruby on Rails at Pune,India. Seating is limited to 15 people. Seat allocation will be strictly on a first-come-first-serve basis. Details below: Date: 21st June thru 25th June 2006 Time: 9:00 am to 6:00 pm Place: SICSR,Pune Address: SICSR,Atur Center,Gokhale Cross Road,Model

Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're

Greetings. I'm teaching linear regression this semester and that means I write more functions for my regression support package "rockchalk". I'm at a point now were some fresh eyes would help, so if you are a student in a course on regression, please consider looking over my package overview document here: http://pj.freefaculty.org/R/rockchalk.pdf That tells how you can grab

Hi list, Does anyone have any advice on the following: Incoming calls to our office come in on a SIP trunk. Since all our offices/desks are in close proximity, we would like just a single phone to ring when a call comes in instead of ringing every person's phone. Currently we've got this working by having all the phones in a callgroup/pickupgroup and incoming calls ring the

Of course, big brother wants his say in the matter. http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead

Here are some notes from the scalability session of last week''s Rails camp. They were entered by another session participant and are posted at: http://www.rubyonrailscamp.com/10%3A15%2Bsession%2B-%2Bscaling The key points from my point of view: - the Ruby VM is sketchy, rather like the Java VM around 1997 - the single threaded nature of Rails dispatch handling means we may incur a

Nik, I'm not sure that "NOP" is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have "OK" under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. As for asterisk, here's what I get

Hello, I'm using asterisk queues, for reception phone, and I have small problem: I have only one phone as queue member, and the problem is, that ALL channels waiting in queue are ringing on it. And if there are too many people ringing on it, it's not possible to use attended transfer then... Is it possible to limit maximum ringing calls from queue? or some other tip? thanks a lot in

Setting up pick/callgroups really tightly is the way to go - one of our techs here does that all the time for clients. PaulH On Wed, 2007-06-20 at 20:51 +0930, Tom Lanyon wrote: > Hi list, > > Does anyone have any advice on the following: > > Incoming calls to our office come in on a SIP trunk. Since all our > offices/desks are in close proximity, we would like just a

Hi all, Having fixed my problems with the call waiting ringing on the GS phones, I needed to extend that with a campon facility (available on some legacy systems - sort of callwaiting without phone ringing). I've managed to implement that by adding/modifying app_queue.c. Basically, when calling the SIP phone, and if busy, I can camp the call onto that extension, with MOH, allowing the caller