[Asterisk-Users] Asterisk Newbie

Henry Devito hdevito at mchsi.com
Tue Mar 15 10:13:16 MST 2005


1 Transcoding is between codecs.  ulaw to g.729 for example

2  I prefer AMP but unless you install it with *@home it could be a pain.

3.  You need a clock source for meetme and other features to work so if you don't have any digium hardware you must use ztdummy

4.  Unless you are using a VPN or STUN you must open the port in your firewall manually

In addition if you search on google by 'list.digium.com : whatever subject'  you could find the answers to a lot of questions.

Have a good day,  I hope this helps you get on your way.

Henry
  ----- Original Message ----- 
  From: Fabian Borot 
  To: asterisk-users at lists.digium.com 
  Sent: Tuesday, March 15, 2005 10:56 AM
  Subject: [Asterisk-Users] Asterisk Newbie


  Hello all
  I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no  a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong.

  That is why I will post my questions here:
  1- Transcoding: is this when you go from g711 to g729 for example? Or when you go from SIP to IAx?
  2- What is the best GUI tool to configure  * ?
  3- Do I need to install a PCI (fxo or fxs) to have meetme, music onnhold etc?
  4- If I have a SIP device behind a firewall the supports SIP transformations (sonicwall pro230) and the * is outside the firewall, do I have to open ports 5060 anyway?
  What about the audio?

  Regards

  Fabian



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