Re: [Asterisk-Users] Now that Nufone is dead...

2006-05-23 Thread Erwin de Raad
From: Carlos Chavez [EMAIL PROTECTED]

  Now that Nufone is dead, what are other providers of 800 numbers that
 work with Asterisk?


Not entirely dead. Yesterday I received an e-mail requiring me to push some
buttons on their dashboard. I think they are still trying.
I switched to exgn.net.  They have ready to use * configs and US+Canada
toll-free. To my surprise they even send e-mails when a call doesn't go
through.

Erwin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] OT: SIP aware firewalls?

2006-01-05 Thread Erwin de Raad
- Original Message - 
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 05, 2006 5:33 PM
Subject: RE: [Asterisk-Users] OT: SIP aware firewalls?


  Until now I've only used IAX2 to connect to ITSPs. I've been
  toying with a SIP connection to Gizmo Project, but not yet
  successfully. It brings to mind a question. At what point
  does it make sense to consider a SIP-aware firewall such as
  those from Ingate?

 You should be able to run SIP through m0n0wall quite happily - we have a
 number of client sites with SIP phones offsite which connect to the *
server
 behind a m0n0wall box. You'll need to allow 5060 (UDP) for SIP, then an
 appropriate port range (as definted in rtp.conf) for the RTP streams.

 You'll obviously also need to apply any QoS rules to both the SIP and RTP
 streams.


Totally agree. I moved from Kerio WinRoute (claims to be SIP aware  not) to
Monowall and all SIP/NAT issues went away.
It doesn't do QoS but you can do bandwith/traffic shaping which also should
work fine.

Erwin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] OT: SIP aware firewalls?

2006-01-05 Thread Erwin de Raad
- Original Message - 
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 05, 2006 5:33 PM
Subject: RE: [Asterisk-Users] OT: SIP aware firewalls?


  Until now I've only used IAX2 to connect to ITSPs. I've been
  toying with a SIP connection to Gizmo Project, but not yet
  successfully. It brings to mind a question. At what point
  does it make sense to consider a SIP-aware firewall such as
  those from Ingate?

 You should be able to run SIP through m0n0wall quite happily - we have a
 number of client sites with SIP phones offsite which connect to the *
server
 behind a m0n0wall box. You'll need to allow 5060 (UDP) for SIP, then an
 appropriate port range (as definted in rtp.conf) for the RTP streams.

 You'll obviously also need to apply any QoS rules to both the SIP and RTP
 streams.


Totally agree. I moved from Kerio WinRoute (claims to be SIP aware  not) to
Monowall and all SIP/NAT issues went away.
It doesn't do QoS but you can do bandwith/traffic shaping which also should
work fine.

Erwin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Toll Free Providers

2005-12-20 Thread Erwin de Raad
- Original Message - 
From: John Reynolds
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Tuesday, December 20, 2005 4:13 AM
Subject: Re: [Asterisk-Users] Toll Free Providers


I have nufone.net for 800 and all seems fine... although my useage is very
low.  2 cents a min.

JR


I agree.
I have a 866 DID connected to * in NL and we are very satisfied with the
quality of the connections.

Erwin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Re: Does hardware like this exist...?

2005-12-17 Thread Erwin de Raad
 BJ Weschke wrote:
  On 12/16/05, Evert Meulie [EMAIL PROTECTED] wrote:
 
 Hi all!
 
 I am looking for a device that I can stick in a USB-port on my Asterisk
server and that allows me to connect one/more (cordless) PSTN-phones in such
a way that they'll work with SIP/Asterisk. I know
 there are USB-phones, but what I'm looking for is 'the USB-phone without
the phone', if you know what I mean...   ;-)
 

Around Christmas last year I ordered a VTA1000 from
http://www.pcphoneline.com
It uses a Windows app so not sure (or tried...) how this works on linux.

Mine is the VTA1000-Skype and I tied it to a SPA3000. This is my Skype to *
gateway.

Erwin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] :: Strange way of receiving calls ::

2005-04-02 Thread Erwin de Raad
With AMP I managed to set up a group which rings with an incoming POTS
call.  With AMP also, I have also managed to create a Digital Receptionist
BUT this requires caller input, which is not what I need :(

Any help would be appreciated! :) Thanks to you all

With AMP you are limited in what you can do through the GUI.
Try selecting '1' at possible options and use 't' (timeout), So no user
input is required. The phones will ringer AFTER your message has completed.
Your caller hears the ringtone, no music.
I don't know where the default timeout value is defined.

Anything further than that is to be hardcoded in extensions.conf

Take care,
Erwin.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] install BRIstuff on *@home?

2005-02-23 Thread Erwin de Raad
From: Niksa Baldun [EMAIL PROTECTED]
 Each release of BRIstuff is made for a specific * version. BRIstuff
 installer automatically downloads the correct version, patches and
 installs it. You should just run the install.sh and it will replace your
 current * installation. Your existing configuration (extensions.conf
 etc.) will not be changed.
 
 Bear in mind that you have to copy the BRI modules (qozap.ko and
 zaphfc.ko) manually to your /lib/modules/`uname -r`/misc directory.
 
 Erwin de Raad wrote:
 
  I'm still trying to install a HFC-s BRI card onto [EMAIL PROTECTED]

Hi Niksa,
Thank you for your response.
I will go ahead then with the install.
I'm deducting what cannot be wrong when it doesn't work...  ;-)

Cheers!
Erwin.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] grandstream budgetone-100 updates

2005-02-23 Thread Erwin de Raad





- Original Message - 
From: Colin Anderson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, February 23, 2005 7:48 PM
Subject: RE: [Asterisk-Users] grandstream budgetone-100 updates



  Can anyone tell me why these fail each time?

 I can, but I won't since this is a FAQ - google it!

  Also what is the latest revision?

 goto www.grandstream.com

 See, this is exactly what I'm talking about. How churlish. Why did he even
 reply? I have a BT100 sitting on my desk, and it took me all of 10 seconds
 to find out what the rev was.

 Dean: Latest published firmware is 1.0.5.16. Dunno about TFTP problems,
 though. Maybe bad checksum? Try to re-extract from a new copy.



Have a look at http://gs-firmware.gratissip.dk/  They have the .22 rev.
Seems to be OK.
They also have http update, just put
gs-firmware.gratissip.dk/firmwares/latest
in your budgetone and set tftp to all '0's
Works for my 101's.

With kind regards, / Met vriendelijke groet,
Erwin de Raad.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] install BRIstuff on *@home?

2005-02-22 Thread Erwin de Raad



I'm still trying to install a HFC-s BRI card onto 
[EMAIL PROTECTED] .6
I'm new to this so I probably am overlooking the 
obvious.
Can I just install BRIstuff onto a fresh [EMAIL PROTECTED] install?

The BRIstuff installer downloads another * from 
Digium. Will this interfere with the @home install and must I comment out the 
Asterisk install in the BRIstuff install.sh file?

Any pointers are much appreciated.

Regards,
Erwin.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] HFC-S ISDN card on *@home

2005-02-20 Thread Erwin de Raad
It seems so simple, but I'm having no luck installing a HFC-s ISDN BRI card
on [EMAIL PROTECTED] 0.5.
I probably have to install BRI-stuff from Junghanns.net but that also
downloads and installs another copy of * from Digium.

I'm not sure if zaphfc has to be installed *before* Asterisk or if it's OK
to do this afterwards.

I've seen this question before, but: Anyone successfully installed a HFC-s
card on [EMAIL PROTECTED] Please post the steps you had to take. I'm sure quite 
some
list-members are interested!

With kind regards
Erwin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] help with @home

2005-02-20 Thread Erwin de Raad
Message also do you have to use a ip phone to record your greeting because
this wav file stuff isn't working.

I didn't try uploading. You can just setup a SIP softphone and dial *77 when
looking at the menu you want to record in the GUI.

Regards,
Erwin

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] help with @home

2005-02-20 Thread Erwin de Raad
Message I'll buy a IP phone tomarrow so i can do that

No need:
http://www.xten.net/index.php?menu=productssmenu=download

Regards,
Erwin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users