similar to: h.323 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

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

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

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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

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

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'

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",

Hi This is the output from show dialplan dial-sipmnf-sippt-pstn [ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ] 's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]

Anyone using the UIP200 with *? I am having difficulty getting the phone to register and *stay* registered for more than 4 seconds. The * console shows the UIP200 registering and records the user agent. The UIP200 displays station name and time on its LCD, then 4 seconds pass and it shows #1 DISCONNECTED. The * console reports no problem and believes the device to still be registered. A call

My users are reporting that some voicemail messages are being cut off in the middle of being played back. The recordings are OK (they play fine when forwarded to e-mail, and they can often be accessed OK during a later call to voicemail). I found nothing in the archives on this -- ideas anyone? RH9, P4, CVS-HEAD-09/02/04-08:44:34, aggressive echo suppression turned on. Jim Shilliday IT

I have a dilemma. I am trying to setup an asterisk setup for about 10 people, but with the ability to expand to 100s. We are looking into hardphones to use for our systems. Here are the requirments I am looking for: SIP Not POE Full duplex Speakerphone $100-$300 We would like to have the cordless and corded option if possible (at least for 3 of the 10 people). So here are the

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

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

I have a box running asterisk 1.4.17 that had been working. it has 2 uniden phones connected on it. This was working and now the phones dont ring when calling each other. below is the sip debug. I cant see why the other phone does not ring? I also tried changing the canreinvite for no to yes but that made no difference after restarting. Very simple network. server, linksys router and 2 phones.

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, #