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List doesn't seem to be posting out - still active here http://lists.digium.com/pipermail/asterisk-users/2005-June/date.html but not being received by email (time warner is the isp but other emails coming in every few minutes as per normal). Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

I have a client who works from three locations, he has a Polycom 500 at each location connected to the same PBX. How would you configure the phones? Is it practical to have the extension follow him, I don't want to rely on him turning the phone off when he leaves each location, but he does want to maintain the same extension and voicemail. Chris Mason

...ress="203" reg.2.auth.userId="blah" reg.2.auth.password="blah" reg.2.server.1.address="192.168.1.8" reg.2.server.1.port="5060"/> </PHONE_CONFIG> - Noah > On Wed, 2005-08-31 at 16:36 -0600, Andres Paglayan wrote: > >> Hi, >> >> I am having problems on getting the second line to work on a >> Polycom 301, >> >> this is the phone.cfg file, >> the * box is 192.168.1.8 and the phone is 192.168.1.18 >> I am not 100% sure about what the reg.x.address shou...

Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass,

Hello everyone, I am racking my brain with a modelling question and can''t wrap my around the benefits of STI or Polymorphisms. I am trying to model Questions and Answer behavior but I think I am not using rails full power here. I have (in simple terms) Question has_one Answer Answer belongs_to Question I also have Topic has_many Posts Post belongs_to Topic Now i am

Hi List; I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? By the way: How can I know the asterisk and zaptel version extactly that I compiled them? In other words, asterisk 1.4.... and zaptel 1.4.... ? Regards ------------- ITS IP Telephony and Contact Center Engineer Eng.

I'm experimenting with realtime (CVS HEAD), but using odbc to a third-party database (progress) instead of mysql. Following the instructions on voip-info, I created a table for voicemail called rtvm with the following fields: CREATE TABLE `rtvm` ( `uniqueid` int(11) NOT NULL auto_increment, `customer_id` int(11) NOT NULL default '0', `context` varchar(50) NOT NULL default

Hello list, I am getting the list with days of delay, take for example this message: Received: from unknown (HELO lists.digium.com) (216.207.245.17) by mxavas16.fe.aruba.it with SMTP; 29 Jun 2007 13:38:37 -0000 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from <asterisk-users-bounces at lists.digium.com>) id

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Hi, this is to inform everybody that the translation of my new book (unstable version) is online at http://www.the-asterisk-book.com The book is a GNU FDL project. So everybody who wants to participate is welcome to do so. Also, everybody who needs material for his own work, feel free to take it as long as the new material will become GNU FDL too. I am glad that Stephen Bosch (who you

Hello, How would you implement free sitting ? The idea is to offer teachers the ability to share the same desk and hardphone : for instance, Mr Foo is teaching mechanics on mondays while Mr Bar is teaching english on wednesdays. Each has his own extension but use the same hardphone. 1. Does a program check a calendar or database somewhere to allocate a phone to a user (as teachers schedules are

Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the

Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting

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Hi, I have an X100P which receives an analog line from another PBX. These are the relevant entries in extensions, PHONE1=Zap/1 [macro-extensions] exten => s,1,Dial(${ARG1},20) exten => s,2,Voicemail2(u${ARG2}) exten => s,102,Voicemail2(b${ARG2}) exten => s,103,Hangup [home] include=>tozap exten => 2201,1,Macro(extensions,${PHONE1},${PHONE1VM}) exten =>

Hi, My users are currently using a console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and

Hi All, I have a (legacy) Praxton PBX, it has a PRI T1 input card and 64 analog extensions through 4 amphenol connectors. We receive 12 voice channels (other 12 are idle) and have 100 DIDs. No caller ID thru PRI though. The Praxton box is amazing in terms of configuration and flexibility but has no VoIP support and the company went poof and it is no longer supported, nor spare parts are

Hi, this snippet <%= submit_to_remote(''create_button'', ''Add Phone'', :submit => "phone_form", :url => send( "#{@phonable_type.to_s.downcase.singularize}_phones_path", @phonable_id),