Posted: Fri Sep 24, 2004 9:53 am Post subject: [Asterisk-doc] docs/volume-one vm1chp4-channelconfig.xml,1.6
Comments:
Update of /cvsroot/asterisk/docs/volume-one
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv30610/docs/volume-one
Modified Files:
vm1chp4-channelconfig.xml
Log Message:
blitzrage
- add in Chris Tooley's latest patch (posted to the mailing list on Sept.
24, 2004) after some minor modifications :)
- this patch still needs to be viewed in the PDF, didn't have time since
I'm at Astricon.
Index: vm1chp4-channelconfig.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/volume-one/vm1chp4-channelconfig.xml,v
retrieving revision 1.6
retrieving revision 1.7
diff -C2 -d -r1.6 -r1.7
*** vm1chp4-channelconfig.xml 24 Sep 2004 17:34:34 -0000 1.6
--- vm1chp4-channelconfig.xml 24 Sep 2004 17:54:01 -0000 1.7
***************
*** 31,34 ****
--- 31,35 ----
recieving calls.
</para>
+ <sect2><title>FXS</title>
<para>
If you have a Digium card that can provide an FXS channel,
***************
*** 57,60 ****
--- 58,64 ----
look out for.
</para>
+ </sect2>
+
+ <sect2><title>FXO</title>
<para>
The main difference in the settings that are provided above when configuring
***************
*** 78,82 ****
channel => 1
signalling=fxs_ls
! channel => 2
</programlisting>
</informalexample>
--- 82,86 ----
channel => 1
signalling=fxs_ls
! channel => 2
</programlisting>
</informalexample>
***************
*** 93,96 ****
--- 97,101 ----
switchtype=national, and signalling=fxs_ls.
</para>
+
<para>
We now having a working example of a zapata.conf with an FXS channel (1) and
***************
*** 99,102 ****
--- 104,108 ----
hooked directly to a standard analog telephone.
</para>
+ </sect2>
</sect1>
***************
*** 109,112 ****
--- 115,119 ----
up the IAX communications in your iax.conf file.
</para>
+ <sect2><title>General Settings</title>
<para>
First we need to set up our IAX globally used settings:
***************
*** 126,129 ****
--- 133,138 ----
5036.
</para>
+ </sect2>
+ <sect2><title>Allowing communications</title>
<para>
Unfortunately the settings above are just a start and don't allow us to
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*** 164,167 ****
--- 173,183 ----
</para>
<para>
+ 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>
+ <para>
We have 2 different kinds of connections to IAXTel, the peer and the user.
This allows us to decide that inbound calls can come from one server and
***************
*** 170,183 ****
trunking the servers together.
</para>
</sect1>
<sect1>
<title>SIP</title>
+ <para>
+ 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>
!
! <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.
+ </para>
+
+ </sect2>
</sect1>
</chapter>
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