[asterisk-users] Blacklist to check http://whocalled.us

2006-10-08 Thread Noah Swint
There's a product below on the market that checks whocalled.us to
determine if a telmarketer should get the Zapteller.  Do you know if
that's something that could possibly be included into the blacklist or
in a macro.

http://venotec.com/product/tms/


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Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source

2006-01-03 Thread Noah Swint

Do you have a url for the device?



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On Tuesday 03 January 2006 05:48, Paul Dugas wrote:
 On Mon, 2006-01-02 at 16:06 +, Jonathan Attwood wrote:
  I use a Dock-n-Talk in conjuction with a Sipura SPA3000  Asterisk.

 Does this unit require any funky dialing when placing outbound calls
 from * through the phone?  Do the docs indicate operation is any
 different between CDMA, TDMS, AMPS, or GSM phones?  I'd guess not or, if
 so, it was simple to handle it in the dialplan but I'm curious anyway.
 I've been considering this as a way to have work calls that come to my
 cell appear different to the server.  At the moment, I have my GSM phone
 forward calls to the house when it's off so I can't really tell between
 them.
I have good experience with a GSM-box I've bought from cybertelecom and
SPA3000. GSM-box acts as a Dock-n-Talk because is it allows in and out
dialing. The advantage is that one doesn't need even a mobile phone, but 
only

a SIM card. The whole thing is like porting a number.

There are 2 FXS ports. One could go to an ordinary phone, the other to
SPA3000.

The disadvantage is that you have one more number for your friends to
remember. Otherwise is stable, and as-easy- as-PnP instalation, if you 
don't

forget to disable the pin lock as I did :-)
benchev


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RE: [Asterisk-Users] X100P troubles?

2005-11-13 Thread Noah Swint

Are you running off the rpms  or compiled version?


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Re: [Asterisk-Users] Asterisk as Media Gateway (was: ATT CallVantage Asterisk

2005-07-03 Thread Noah Swint
I am interested.  I'm using Callvantage and Centillium TA.  I'm also  using 
Asterisk with a X100P Card.  Eventually I want to replace this with my 
Linksys NSLU2.  I'm having mystery power issues with my linux server.  
Currently I have calls from soft and ip phones going to asterisk then out to 
the ta fxo to fxs. Regular pots phones go to a telephone distribution panel 
with the Centillium bringing service to line one.  I'm interested in using 
the MGCP  hopefully to be able to use sip outside of my lan.




Edwin Horton edhorton at navcorp.com  wrote:

I noticed that there is some interest in MGCP slave operation for Asterisk
to enable it to work with the ATT Callvantage offering.  I have tried the
FXS/FXO connection to Asterisk and the Linksys TA with little success.
Dropped calls are the biggest problem, which does not occur when the phone
is directly connected to the TA.  Anyway, I am very interested in working on
this project if others are still interested.

Ed Horton


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[Asterisk-Users] Asterisk with regular analog phones

2005-02-25 Thread Noah Swint
Can regular analog phones be used and act as extensions, or does an fxs
device need to be put into place. I saw this on voip-info.org.  How
would extension setup be possible without the fxo being aware of the
name of the device?


for Analogue Phones connect to Zaptel
http://www.voip-info.org/wiki-Asterisk+zap+channels channel

See Asterisk vertical service activation codes
http://www.voip-info.org/wiki-Asterisk+vertical+service+activation+codes
for ZAP channels

* *Call Hold*: Normally implemented by your phone
* Unattended Transfer (or blind transfer)
* *Consultation Hold*: Normally implemented by your phone, for
* *Unconditional Call Forwarding*
* *Attended Transfer* (or consultative transfer): See Asterisk
  tips zap transfer
  http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer
* *No Answer Call Forwarding*: Implemented by yourself in the dial
  plan. See the tips  tricks page
  http://www.voip-info.org/wiki-Asterisk+tips+and+tricks for ideas
* *Busy Call Forwarding*:Implemented by yourself in the dial plan.
  See the tips  tricks page
  http://www.voip-info.org/wiki-Asterisk+tips+and+tricks for ideas
* *Single-Line Extension*:
* *3-way Calling*: Normally implemented by the phone
* *Incoming Call Screening*: Implemented by yourself in the dial plan
* *Find-Me*:
* *Call Pickup*: Supported in the standard installation
* *Outgoing Call Screening*: Implemented by yourself in the dial plan
* *Automatic Redial*: You should be able to implement this in the
  dial plan with some AGI support
* *Manual Redial*
* *Do-not-disturb (DND)*
* *Message waiting (MWI)*: Implemented in Asterisk, but must be
  support on the phone


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