similar to: TDD over Asterisk

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

I thought I'd share my Asterisk experience, which hasn't exactly been as pleasant as I would like but now seems usable in most ways and more then I expected in other ways. I wanted a home PBX system, that would let me treat different callers different ways depending on CID. I initially bought the Digium developer's kit to try things out. That's a single port TDM400 and a X101P.

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

I didn't know if it would work or not, but I figured I'd try slow-speed half-duplex TDD over GSM & Vonage. I called a AGI script I have that speaks to TTYs, by calling from Vonage to one of my Voicepulse lines. I don't control the Vonage codec, so I have no idea what it uses, but I am using GSM for the Voicepulse line. Everything worked fine - echo canceling didn't cause any

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

I can see that Asterisk supports TDD text channels, as there is a tdd.c file in the source, this appears to be exposed via TDD MODE in the AGI interface? I cant find any documentation anywhere on how to use this. Has anyone done this? tdd.c seems to only support 45.5 baud calls. Ideally I'd like a simple TDD answering machine, that sends "PLEASE TYPE MESSAGE" and receives TDD

On voip-info I found a few dated references to TDD support being in the alpha stage and buggy. Can anyone direct me to any newer information on this option? Thanks -- Karl Harris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100616/d18f876f/attachment.htm

[snip good stuff] > After that you realize that TDD just makes you happy, so you start to apply it to other aspects of your life. Wait, I thought happiness had something to do with goats and feeling guilty about it. I''m confused. Again. But I would second Dylan''s suggestion of seeing a video of TDD in action w/ RoR. Having never tried this, it''s difficult to

Here is a blog about TDD. http://hl.thoughtworkers.org/?p=7 It talks about starting TDD from writing functional/acceptance tests according to acceptance criteria of stories, then drilling down to the unit tests. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/rspec-users/attachments/20071017/9bbe579c/attachment-0001.html

I've read the WIKI. Any additional information on the TDD mode? Anyone done a TDD to email app? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

We are working on a TDD application and testing with an Ultratec SuperPrint 4425 in Australia. We've got it to exchange characters fairly cleanly for a minute or two but after that we seem to be getting a signal but no character in Asterisk. Has anybody done any work with the Asterisk TDD functionality and got it to work reliably? Does anyone have any idea what might be causing the

I ditched the idea of using Asterisk straight through my router, there was too much to set up in too little time for me. I've found a spare machine and installed Asterisk@Home on it. Things run smoothly except for connecting to my IAX provider; It doesn't even look like packets are going out at all. Here's my config: ANY Cable ------> Firewall/Router

I connected two plain old telephones to FXS lines of a TDM400P (defined as fxoks in zaptel.conf), one of them dials the other with Dial(Zap/2), I talk to myself for a while, hang up either of the phones, and the phone that remains off-hook gets the congestion tone until it goes on-hook (at least as long as I've cared to wait). I don't have a voltmeter on the line, but if I'm hearing

Hi, I''m learning TDD without using Shoulda or RSpec. When I played with RSpec, there was a tool to show test coverage visually. Is there any similar function built-in to Rails Test::Unit? Or do I need to add it on my own? What''s the recommended tool? I tried `rake -T test`, but didn''t find anything about coverage. -Jon

Mosquito, the TDD framework for Camping has been updated (0.1.3). We''ve tried to keep it backwards compatible, so your apps should continue to work. But feel free to submit a bug report at RubyForge if you have any problems. Changes: http://rubyforge.org/frs/shownotes.php?group_id=351&release_id=14845 Download: http://rubyforge.org/projects/mosquito/ Docs:

Here's something I haven't been able to discover as of yet - I need to set up a "direct link" from my Asterisk box to an external line... basically I need to be able to pick up an internal extension and have it call a local phone number. This is call forwarding, I know - the question that I have is how do I set it up so that the extension always forwards. There will never be a

TDD, SVN and migrations seem to be a natural combination for model development: create a test, add the column with a migration, write the code, and check it all in when you''re done. The trouble is that this ends up creating one migration per field. I guess it''s not horrible in and of itself, but it does have a code smell to it. Has anyone come up with a different way of doing

After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman

Good Day All, I am experiencing some weirdness using the E&M channel and hope you can offer a little assistance with the problem I am having. 1) call comes into channel 25 (Second Span first channel of a Digium Quad PRI from SBC-PRI) 2) Call is sent to channel 1 (First Span first channel on the Digium Quad PRI connecting an ADTRAN via E&M Feature Group D) 3) Between rings one and two

Andrew, I don't think it's a Polycom issue. We took Asterisk out of the picture and had our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session Progress, and the polycom's play the correct tones in this case. We WANT Asterisk to send progress tones in band. In our case it IS needed.