similar to: Dial-plan transfer

Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible? Any help would be great! Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA

We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... " I am sorry that's not a valid extension" before i get a chance to enter anything. Any

I've been looking for a good iax or sip <==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office:

We just got a shiny new Snom 220 it's working great except..... When a call on line one is picked up and another call comes in any attempt to hit the transfer button actually connects the two incoming calls rather than present a dial-tone for transfer. Which is quite a surprise for the people on the other end. Did I miss a setting? anyone have some insight? Matt Hohman New Heights Church

hallo, does anybody successfully managed to get swissvoice ip10s with h323 firmware work with asterisk ? mgcp firmware works fine, but with h323 i'm still getting one way audio. regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

I've been looking out for the Uniden UIP1868 for a while now, but I haven't seen it anwhere that I'm used to buying things from. According to froogle, a couple of places (that I've never heard of) have a small number in stock (small = 10 in this case). I'm doubly suspicious because even uniden's own online store doesn't have them available yet, not to mention

Hi all, I'm experiencing a problem with SIP channels going ZOmBIE after the following sequence of events: - IAX2 client calls SIP client - SIP client consultive transfers (using sip REFER) the call to a MeetMe extension, and hangs up. At this point, the IAX2 client will indeed be in the meetme room, but a 'show channels' at the * CLI reveals that the SIP channels that were involved

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Hi there... I have a simple * setup with about 11 Soft phones (SJ Phone). The clients don't support a consultive or supervised transfer (I believe that's what it is called). Tris is a feature much desired by the powers that be and they want me to "make it work" :) I was wondering if there was a way to do this with and AGI script or the like so that when Staff 1 gets an external

Any ideas how I would go about writing specs for views which make use of content_for? I''d like, for example, to be able to specify that ABC view places XYZ in the sidebar, which I do using content_for(:sidebar). Am I missing something obvious? Kyle

describe Asset, " when destroyed" do fixtures :assets, :videos, :sites, :publish_settings before(:each) do @asset = assets(:test_asset) @mock_hook = mock("hook") @asset.video.stub!(:hook).and_return @mock_hook # error occurs here end it "should call the delete hook" do @mock_hook.should_receive(:update).with("test_video",

What is the correct way to mock out the errors on a Rails model? I''m assuming i need to say @mock_thing = mock_model(Thing) @mock_thing_errors = mock("errors") @mock_thing_errors.should_receive(:full_messages).and_return("An error") @mock_thing.should_receive(:errors).and_return(@mock_thing_errors) Just wanted to check the best practice on this kind of thing and how

Is it possible to boot a no-emulation boot image from ISOLinux? I am particularly interested in having the WinXP Install CD, as one of the boot images in ISOLinux. - Murali ################################################################ # Ry 254, Ryerson Hall, # Ph : +1 773 702-1031 (Off) # # 1100 E 58th Street # +1 773 575-4836 (Cell) # # Chicago, IL - 60637, #

Look into the original XP cd and see where NTLDR is located, then place it on your cd in the same directory and it 'should' work. I have created some 2000 XP boot cd's with integrated recent service pack's and they still boot. But I do that with cdrecord+original boot sector, not with any isolinux stuff... But I hope this help! -----Original Message----- From: G. Murali

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

I am trying to get a new Asterisk installation running using a Zultys ZIP2 phone and a Uniden 200 phone. I have the system working reasonably well (although probably not optimal) except for a couple of items. First, I can't get the voice mail message light to work on the Zultys phone but it works just fine on the Uniden phone. Second, the time presented on the Uniden UIP200 phone is 1 hour

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect'

Hello all, I plan to make an (emergency) multi OS bootable CD which contain: - Win98 boot floppy (create from Win 98 control panel, add remove program) - WinME boot floppy (create from Win 98 control panel, add remove program) - Linux boot disk - Win NT boot disk - Win 2000 recovery console - Win XP recovery console - Winternal Administrative Pak (ERD Commander etc) I am new to Linux and feel