Index: vm1chp4-channelconfig.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/volume-one/vm1chp4-channelconfig.xml,v
retrieving revision 1.4
diff -r1.4 vm1chp4-channelconfig.xml
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> 		<sect2><title>FXS</title>
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> 		</sect2>
> 		<sect2><title>FXO</title>
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> 		</sect2>
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> 		<sect2><title>General Settings</title>
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> 		<sect2><title>Allowing communications</title>
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> 		The firt change you will notice actually comes in the general section.  It
> 		is a line to tell IAXTel that we are here and that calls to that IAX user
> 		should be routed to your asterisk server.  It's like connecting to your IM
> 		(AOL Instant Messenger, Yahoo, MSN, etc) so that when other people send you
> 		a message you get it wherever you are logged in.
> 		<para>
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> 		</sect2>
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< 		Extension 1001 == SIP/1001 ${JOHN}
< 		Extension 1002 == SIP/1002 ${MARY}
---
> 		The Session Initiation Protocol (SIP) is rapidly becoming the most widely
> 		support VoIP protocol.  Like IAX, SIP is pretty easy to set up.  There are
> 		some gotcha's with the protocol though.  Be aware that while your channels
> 		may be set up correctly, SIP does not handle NAT very well, and this can
> 		be a source of significant headaches.
> 		</para>
> 		<sect2><title>General Settings</title>
> 		<para>
> 		The first thing tht needs to be done is setup the general settings.  Much
> 		like IAX this allows you to make settings that all sip connections will use.
> 		</para>
> 		<informalexample>
> 		<programlisting>
> 		[general]
> 		port = 5060                     ; Port to bind to
> 		bindaddr = 10.78.1.90           ; Address to bind to
> 		context = default               ; Default for incoming calls
> 		srvlookup=yes                   ; Enable SRV lookups on outbound calls
> 		dtmfmode=inband
> 		allow=all                       ; Allow all codecs
> 		</programlisting>
> 		</informalexample>
> 		<para>
> 		As you can see, the settings are very similar to IAX.  We have a port, address,
> 		context, and allow.  The srvlookup setting is a way to look up host names.
> 		If set to yes, DNS lookups will happen on SRV records instead of A records to
> 		accomodate for load balancing.  The dtmfmode setting is used to determine how
> 		Asterisk should listen for tones, such as someone dialing an extension.
> 		</para>
> 		</sect2>
> 		<sect2><title>Allowing Communications</title>
> 		<para>
> 		For SIP channels to be used, clients have to be given permission to authenticate
> 		to Asterisk via SIP.  Also for Asterisk to be used as a client (for something
> 		like Free World Dialup phone service via SIP, the client settings must be
> 		setup.
> 		</para>
> 		<para>
> 		<informalexample>
> 		<programlisting>
> 		[general]
> 		port = 5060                     ; Port to bind to
> 		bindaddr = 10.78.1.90           ; Address to bind to
> 		context = default               ; Default for incoming calls
> 		srvlookup=yes                   ; Enable SRV lookups on outbound calls
> 		dtmfmode=inband
> 		allow=all                       ; Allow all codecs
> 		register => FWDNumber:secretpassword@fwd.pulver.com/EXTEN
> 
> 		[fwd.pulver.com]
> 		type=user
> 		username=FWDNumber
> 		secret=secretpassword
> 		host=fwd.pulver.com
> 		nat=yes
> 		canreinvite=no
> 
> 		[fwd.pulver.com]
> 		type=peer
> 		host=fwd.pulver.com
> 		context=default
> 		nat=yes
> 		canreinvite=no
> 		</programlisting>
> 		</informalexample>
> 		<para>
> 		FWD stands for Free World Dialup, a free VoIP to VoIP service that can be
> 		found at http://www.freeworlddialup.com.  Communications between FWD and
> 		IAXTel are allowed and there are instructions on their respective websites
> 		on how to do this inter-service communications.
> 		</para>
> 		<para>
> 		Again, you can see that we've put a register line in the general section to
> 		let the service provider know that we are the correct client for calls that
> 		would be routed to FWDNumber.  You'll also notice that there is a new context
> 		in the example.  The fwd.pulver.com context allows calls to FWD and from FWD
> 		to be handled by Asterisk.
> 		</para>
> 		<para>
> 		The first entry is the information necessary for outbound calls to use the
> 		FWD service as a client.  It is almost identical to the section under IAX
> 		for "type=user".  The second entry is for authenticating inbound calls, to
> 		ensure that we're not getting fake call routing from another source.
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> 		</sect2>