similar to: Termination Provider

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/

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

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

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

Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone.

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

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

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

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 All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Hi all. I`ve installed a X100P on my box and is working well with incoming and outgoing calls as a trunk with one PTSN line. I want to connect the X100P to my Panasonic 616 EASA-PHONE as an internal extension to permit to users to make calls to SIP devices from analog phones, the problem is when I dial the ext number where the X100P is connected I get busy tone. What config I need to change to

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

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

Noob here, my apologies in advance. Recently my employer decide to stop paying for my home POTS line, so I ordered packet8 for a home line instead of another POTS line, since I really dislike my local phone company, and the POTS line without any long distance would cost more than the $20 to packet8 with unlimited US calling. Anyway, this whole thing got me started thinking about VOIP a lot

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the