voicemail.sgml

SIP Express Router, Linux下的SIP代理服务器,小巧实用,开发测试VoIP设备和应用的必备.

<!-- <!DOCTYPE Book PUBLIC "-//OASIS//DTD DocBook V4.1//EN" [ ]> -->
<section>
<title>Introduction</title>
<para>
	The voicemail system provides <application>ser</application>
	with voice announcement and recording capabilities. Voice
	messages may then be mailed to the called user. The system
	relies on <application>ser</application> for implementing
	the <acronym>SIP</acronym> stack and communicate with it
	through <acronym>FIFO</acronym>. It implements the dialog and media
	handling as described in RFC 3264 (An Offer/Answer Model with
	the Session Description Protocol) and RFC 1889 (Real time
	transport protocol) to realize its goal.
</para>
</section>
<section>

<title>Advantages</title>
<para>
	<itemizedlist>
		<listitem>
			<para>
		            Anyone deploying <application>ser</application> and
			    <acronym>VoIP</acronym> should profit from this 'ready-to-run'
			    application. It plugs into <application>ser</application> as
			    easy as configuring the database location, announce file path
			    and SMTP server address. 
			</para>
	        </listitem>
	        <listitem>
			<para>
			    Further,
			    <application>voicemail</application>
			    integrates the most popular free codecs
			    (G.711ulaw, G.711alaw and GSM 06.10) and
			    its own SMTP client, which means that you
			    don't need to install anything else as
			    <application>ser</application> and
			    <application>voicemail</application>. 
			</para>
	        </listitem>
	        <listitem>
			<para>
			    If you want your voicemail system to support
			    other codecs, a simple plugin system with
			    SDK allows you to integrate them fast and
			    simply (see the basis plugins for examples). 
			</para>
	        </listitem>
	</itemizedlist>
</para>
</section>
<section>

<title>Technical limitations</title>
<para>
	<itemizedlist>
		<listitem>
		    <para>
		            The sound conversion engine doesn't
			    support yet re-sampling. It means that
			    input and output files have to be
			    compatible with the sampling rate of the
			    codec. All codecs included with the
			    distribution work at 8kHz, which means
			    that all the input and output files MUST
			    be sampled at the rate of 8kHz.
		    </para>
		</listitem>
		<listitem>
		    <para>
		            At the moment, voicemail only support the
			    Microsoft Wav file format with PCM 16 bit,
			    Mu-law and A-law 8 bit encoding.
		    </para>
		</listitem>
	</itemizedlist>
</para>
</section>
<section>

<title>Compilation and installation</title>
<para>
        <itemizedlist>
		<listitem>
			<para>
			First, you need to compile Ser with voicemail
			support. Therefore, you must edit Ser's
			Makefile.defs file and uncomment the line with
			'-DVOICE_MAIL' and '-D_TOTAG'.
			</para>
		</listitem>
		<listitem>
			<para>
			Then do 'make all' in Ser's root directory.
			</para>
		</listitem>
		<listitem>
			<para>
			Configure Ser to fit your needs. You can
			report to voicemail example config file to
			know what your configuration file should
			include. Note that voicemail
			only needs to know the user database location
			in order to work. Report to the README file in
			the vm module directory for description of the
			functions and variable that are used by
			voicemail and how they work.
			</para>
		</listitem>
		<listitem>
			<para>
			Finally, compile the voicemail application:

			<programlisting>
			[~/voicemail]$ cd ortp-0.5.0
			[~/voicemail/ortp-0.5.0]$ ./configure
			[~/voicemail/ortp-0.5.0]$ make all
			[~/voicemail/ortp-0.5.0]$ cd ..
			[~/voicemail]$ cd plug-in/gsm/gsm-????
			[~/voicemail/plug-in/gsm/gsm-????]$ make all
			[~/voicemail/plug-in/gsm/gsm-????]$ cd ../..
			[~/voicemail]$ make all
			</programlisting>

			You can then start voicemail with following
			command <command>ans_machine</command> and
			look if the default fit your needs. If not,
			type <command>ans_machine -h</command> to see
			how to change the default parameters.

			  <!--<note>-->
			  If <application>ans_machine</application> is
			  not started or can't be joined while
			  <application>ser</application> tries to
			  communicate with it, the caller will become
			  a '500 internal server error' with a comment
			  saying what the trouble is.
			  <!--</note>-->
			</para>
		</listitem>
	</itemizedlist>
</para>
</section>

<section>
<title>Example ser Config File</title>
<para>
	<example>
                <title>Example ser Config File</title>
		<programlisting>
&voicemailcfg;
		</programlisting>
	</example>
</para>
</section>
<section>

<title>Availability, report bugs, contact the author</title>
<para>
        Ser's Voicemail's home page is hosted at
	http://sems.berlios.de. A snapshot may be downloaded directly
	from the CVS tree. A pre-configured version of 
	<application>ser</application> including
	<application>voicemail</application> will be soon available
	(from version 0.8.11). Bugs can be reported at the voicemail's
	home page. If you want to contact the author, use the contact
	email at the home page.
</para>
</section>