Merged revisions 58931 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | 13 lines

Merge changes from svn/asterisk/team/russell/LaTeX_docs.

* Convert most of the doc directory into a single LaTeX formatted document
  so that we can generate a PDF, HTML, or other formats from this
  information.
* Add a CLI command to dump the application documentation into LaTeX format
  which will only be include if the configure script is run with 
  --enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included.  However, you
  can simply run "make asterisk.pdf" to generate it yourself.  We may include
  it in release tarballs or have automatically generated ones on the web site,
  but that has yet to be decided.

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58932 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 0 insertions, 67 deletions

diff --git a/doc/iax.txt b/doc/iax.txt
deleted file mode 100644
index 9a17c098d..000000000
--- a/doc/iax.txt
+++ /dev/null
@@ -1,67 +0,0 @@
-Inter-Asterisk eXchange Protocol
-================================
-
-Usage:
-======
-The format for the dialing string on Asterisk is:
-IAX/[user@]peer[:exten[@context]]
-
-(Note, []'s denote optional fields).  The peer is either an IP address
-or a peer as specified in the /etc/asterisk/iax.conf file.  Exten is
-an optional requested extension (otherwise "s" will be used), and 
-"context" is an optional context to request.  The user is an optional
-username specified in the peer's iax.conf.  If the user is not specified,
-the peer will select one.
-
-The peer uses a score to determine the best user entry to match against if
-one is not specified:
-
-1. User entry last specified in iax.conf (this is the baseline).
-2. User entry with secret specified and ACL specified.
-3. User entry with no secret specified and no ACL specified.
-4. User entry with no secret specified and ACL specified.
-5. User entry matched via username.
-
-The higher the score the better it is with 5 being an exact match and the maximum
-score possible.
-
-Protocol and rationale:
-=======================
-IAX is a simple, low overhead and low bandwidth VoIP protocol designed to 
-allow multiple Asterisk PBX's to communicate with one another without
-the overhead of more complex protocols like H.323.  Payload is sent with
-a header overhead of only 4 octets.  Control functions (and one payload packet
-per minute or so) is sent with a more complex header of 12 octets.
-
-IAX is slightly stateful.
-
-IAX contains two kinds of packets:  The full header packet type, which 
-contains much information about the frame, in addition to its contents,
-and the mini header type, which is used only for non-reliable voice
-packet delivery.
-
-All packets are immediately transmitted.  Packets are received, but not
-delivered to the actual channels until a given time quantum has passed, in
-order to try to eliminate jitter.
-
-All full header packets must be ackd (except, obviously for the ACK packets
-themselves and not so obviously for hangup packets).  The "timestamp" field of
-ack packets is not the normal offset, but rather a quote of the timestamp as
-included with the original packet that you're acking, and likewise the
-seqno field is the seqno of the packet you're acking, not your own seqno,
-and you do not increment your own sequence number.  ACKing is based on the
-sequence number.
-
-See iax.h for a description of the frame formats. 
-
-IAX internal frames use the AST_FRAME_IAX type.  The subclass of these
-frames is the IAX control number, as seen in iax.h.  The first frame sent
-must be an AST_FRAME_IAX with the control AST_IAX_CONTROL_NEW.  
-
-The AST_IAX_CONTROL_NEW establishes a new connection.  
-
-The first frame sent MUST be an AST_CONTROL_NEW to start a connection.
-
-IAX connnections may require authentication using either simple plaintext
-passwords or an md5 challenge/response pair.
-