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<html xmlns="http://www.w3.org/1999/xhtml"><head><meta http-equiv="Content-Type" content="text/html; charset=UTF-8" /><title>Configuring SIP Telephones</title><link rel="stylesheet" href="styles.css" type="text/css" /><meta name="generator" content="DocBook XSL Stylesheets V1.69.1" /><link rel="start" href="index.html" title="Asterisk™: The Future of Telephony" /><link rel="up" href="asterisk-CHP-4.html" title="Chapter 4. Initial Configuration of Asterisk" /><link rel="prev" href="asterisk-CHP-4-SECT-6.html" title="Configuring an FXS Channel for an Analog Telephone" /><link rel="next" href="I_sect14_tt599.html" title="Connecting to a SIP Service Provider" /></head><body><div class="navheader"><table width="100%" summary="Navigation header"><tr><th colspan="3" align="center">Configuring SIP Telephones</th></tr><tr><td width="20%" align="left"><a accesskey="p" href="asterisk-CHP-4-SECT-6.html">Prev</a> </td><th width="60%" align="center">Chapter 4. Initial Configuration of Asterisk</th><td width="20%" align="right"> <a accesskey="n" href="I_sect14_tt599.html">Next</a></td></tr></table><hr /></div><div class="sect1" lang="en" xml:lang="en"><div class="titlepage"><div><div><h2 class="title" style="clear: both"><a id="asterisk-CHP-4-SECT-7"></a>Configuring SIP Telephones</h2></div></div></div><p>The<a id="ch04_sipconfig" class="indexterm"></a> Session Initiation Protocol (SIP),<sup>[<a id="id4117935" href="#ftn.id4117935">57</a>]</sup> commonly used in VoIP phones (either hard phones, or
softphones), takes care of the setup and teardown of calls, along with any
changes during a call such as call transfers. The purpose of SIP is to
help two endpoints talk to each other (if possible,
<span class="emphasis"><em>directly</em></span> to each other). The SIP protocol is simply a
signaling protocol, which means that its purpose is only to get the two
endpoints talking to each other, and not to deal with the media of the
call (your voice). Rather, your voice is carried using another protocol
called the <a id="I_indexterm4_tt508" class="indexterm"></a><a id="I_indexterm4_tt509" class="indexterm"></a>Real-Time Transport Protocol (RTP; RFC 3550) to transfer
media directly between the two endpoints.</p><div class="note" style="margin-left: 0.5in; margin-right: 0.5in;"><h3 class="title">Note</h3><p>We use the term <span class="emphasis"><em>media</em></span> to refer <a id="I_indexterm4_tt510" class="indexterm"></a>to the data transferred between endpoints and used to
reconstruct your voice at the other end. It may also refer to music or
prompts from the PBX.</p></div><p>In the world of SIP, we call our endpoints user agents, of which
there are two types: client and server. The<a id="I_indexterm4_tt511" class="indexterm"></a> <span class="emphasis"><em>client</em></span> is the endpoint that generates
the request, and the <span class="emphasis"><em>server</em></span> processes the request and
generates a response. When an endpoint wishes to place a call to another
endpoint (such as our softphone calling another softphone), we generate
our request and send this to a SIP proxy.<sup>[<a id="id4118046" href="#ftn.id4118046">58</a>]</sup> A proxy server will take the request, determine where the
request is destined for, and forward it on. Once the two user agents have
negotiated a successful call setup, the media is transported via the RTP
protocol and sent directly between the two user agents. SIP proxies do not
handle media; they simply deal with the SIP packets.</p><p>Asterisk, on the other hand, is called a<a id="I_indexterm4_tt514" class="indexterm"></a> Back-To-Back User Agent (B2BUA). This means that Asterisk
acts like a user agent in either the server (receiving) or client
(sending) role. So when our softphone dials an extension number, the call
is set up between the softphone and Asterisk directly. If the logic we’ve
built into Asterisk determines that you mean to call another user agent,
then Asterisk acts as a user agent client and sets up another connection
(known as a channel) to the other phone. The media between the two phones
then flows directly through Asterisk.<sup>[<a id="id4118104" href="#ftn.id4118104">59</a>]</sup> From the viewpoint of the phones, they are talking with
Asterisk directly.</p><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4118129"></a>Basic SIP Telephone Configuration in Asterisk</h3></div></div></div><p>Configuring a SIP phone to work<a id="I_indexterm4_tt515" class="indexterm"></a> with Asterisk does not require much. However, because
there are so many options possible in both Asterisk and the
configuration of the particular telephone set or softphone, things can
get confusing. Add to this the fact that similar things can have
different names, and you have a recipe for frustration. What we are
going to do, therefore, is give you the bare-bones basics. If you follow
our advice, you should be able to get the sets we cover working (and be
well on your way to getting a phone that we have not covered to work as
well). We are not saying that this is the best way, or even the right
way, but it is the simplest way, and from a working foundation, it is
much easier to take a basic configuration and tweak things until you get
the solution you need.</p><div class="note" style="margin-left: 0.5in; margin-right: 0.5in;"><h3 class="title">Note</h3><p>Just as we did with the <span class="emphasis"><em>extensions.conf</em></span>
file; run the following commands in your bash shell:</p><a id="I_programlisting4_tt516"></a><pre class="programlisting">
<strong class="userinput"><code># mv sip.conf sip.conf.sample
# touch sip.conf</code></strong>
</pre></div><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="id4118185"></a>Defining the SIP device in Asterisk</h4></div></div></div><p>If you put the following in a <span class="emphasis"><em>sip.conf
</em></span>file, you<a id="I_indexterm4_tt517" class="indexterm"></a> will be able to register a phone to the system.
<a id="I_programlisting4_tt518"></a></p><pre class="programlisting">[general]
[1000]
type=friend
context=phones
host=dynamic</pre><p>Not pretty, not secure, not flexible, not fully-featured, but
this will work.</p><p>Even though we have named this SIP device
<code class="literal">1000</code>, and we are probably going to assign it that
extension number, you should note that we could have named it whatever
we wanted. Names such as <code class="literal">mysipset</code>,
<code class="literal">john</code>, <code class="literal">0004f201ab0c</code> are all
valid, popular, and may suit your needs better.<sup>[<a id="id4118250" href="#ftn.id4118250">60</a>]</sup> What we are doing is assigning a unique identifier to a
device, which will form part of the credentials when a call is placed
using the SIP channel.</p><p>Since we want to be able to both send calls to the softphone and
allow the client to place calls, we have defined the <code class="literal">type</code> as <code class="literal">friend</code>. The other two types are <code class="literal">user</code> and <code class="literal">peer</code>. From the viewpoint of Asterisk, a
<code class="literal">user</code> is configured to enter the
dialplan, and a <code class="literal">peer</code> is created for
calls leaving the dialplan (via the <code class="literal">Dial()</code> application). A <code class="literal">friend</code> is simply a shortcut that defines
both a user and a peer. If in doubt, define the type as
<code class="literal">friend</code>.</p><p>The <code class="literal">host</code> option is used to
define where the client exists on the network when Asterisk needs to
send a call to it. This can either be defined statically by defining
something like <code class="literal">host=192.168.1.100</code>,
or if the client has a dynamic IP address, then we set <code class="literal">host=dynamic</code>. When the <code class="literal">host</code> option is set as <code class="literal">dynamic</code>, and the client is configured to
register, Asterisk will receive a REGISTER packet from the endpoint
(i.e., telephone set or softphone), telling Asterisk which IP address
the SIP peer is using.</p><p>If you do not trust your network, you should probably also force
the use of a password by adding the following to the device
definition. This is one of those things that is not technically
necessary, but is probably a good idea:</p><p>
<a id="I_programlisting4_tt519"></a></p><pre class="programlisting">secret=guessthis</pre><p>
</p></div></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4118375"></a>Configuring the Device Itself</h3></div></div></div><p>In the configuration menus of the phone itself (which could be via
a web GUI, through menus on the phone itself, or possibly using
configuration files that are stored on a server), the unique identifier
(which in this case is <code class="literal">1000</code>) again forms part of the
credentials for the authentication process. Naturally, for a connection
to be successful, this has to match in both Asterisk and in the set
itself. The fun begins when you realize that there is no set rule as to
what this identifier is formally called. We have elected simply to call
it the Unique Identifier. </p><div class="note" style="margin-left: 0.5in; margin-right: 0.5in;"><h3 class="title">Note</h3><p>In the<a id="I_indexterm4_tt520" class="indexterm"></a> SIP RFC (<a href="http://www.faqs.org/rfcs/rfc3261.html" target="_top">http://www.faqs.org/rfcs/rfc3261.html</a>),
section 19.1 calls this user token, “the identifier of a particular
resource at the host being addressed,” verbiage consistent with our
usage of <code class="literal">[1000]</code> as the set
identifier in the <code class="filename">sip.conf</code> file
of Asterisk.</p></div><p>Instead, you will want to look for fields that are labeled
<code class="code">user name</code>, <code class="code">auth name</code>, <code class="code">authentication</code> <code class="code">name</code>, and so on.
The thing to remember is that since you know that the Asterisk end of
the equation is configured simply and correctly, you can experiment with
the phone setting until you find a combination that works. This is much
better than the usual suffering that new users go through, as they
change settings in both places and have no luck getting a phone to
register.</p><div class="note" style="margin-left: 0.5in; margin-right: 0.5in;"><h3 class="title">Note</h3><p>We’re gonna say it again: configure
<span class="emphasis"><em>sip.conf</em></span> in the simplest manner possible, and
then don’t change your Asterisk configuration. Trust us; what we have
written here will work. Get your set working (i.e., where you can make
and receive calls), and you will be in a far better position to begin
experimenting with different settings. We have seen too much suffering
(including our own), and we want it to end.</p></div><div class="sidebar"><a id="I_sidebar4_tt521"></a><p class="title"><b>Simplifying sip.conf</b></p><p>The <span class="emphasis"><em>sip.conf</em></span> file (which was copied to the
<span class="emphasis"><em>/etc/asterisk</em></span> directory by the <code class="literal">make
samples</code> command we ran in the previous chapter) contains a
large number of options and documentation inside it, but the file is
actually very minimal if you remove all the commented parameters. The
default file really breaks down to just the following few lines being
uncommented by default:</p><p>
<a id="I_programlisting4_tt522"></a></p><pre class="programlisting">[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will
; listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
[authentication]</pre><p>
</p><p>The <code class="literal">[general]</code> section
contains the<a id="I_indexterm4_tt523" class="indexterm"></a> options that will apply to all SIP clients and trunks.
Some settings elsewhere are set only in the <code class="literal">[general]</code> section, and others can be set in
the <code class="literal">[general]</code> section as defaults
for all conditionals unless overridden. These options are listed under
the two columns labeled <code class="literal">[users]</code> and
<code class="literal">[peers]</code> below the <span class="keep-together"><code class="literal">[authentication]</code></span> header.</p><p>Generally, the commented-out options will show you the default
setting Asterisk uses, or will tell you the default option in the
option’s description.</p><p>You can also check the current state of the SIP channel in
Asterisk with the <code class="literal">sip show settings</code>
CLI command.</p></div><div class="note" style="margin-left: 0.5in; margin-right: 0.5in;"><h3 class="title">Note</h3><p>If you are running Asterisk and a softphone<a id="I_indexterm4_tt524" class="indexterm"></a> on the same system (i.e., running an X-Lite softphone
and Asterisk on a laptop or desktop), then you will need to modify the
SIP port that client listens on. It will need to be changed from 5060
to 5061 (or some other unused port) so that Asterisk and the softphone
do not interfere with each other.</p></div></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4118615"></a>Essential Server Components</h3></div></div></div><p>Before we get <a id="I_indexterm4_tt525" class="indexterm"></a>into how to define these files, there are a few things
that need to be configured on your server. Running the right services on
your network will ensure your<a id="I_indexterm4_tt526" class="indexterm"></a> Polycom sets can autoconfigure from the moment you plug
them in. There’s a little work involved here, but we promise that the
payoff is worth it. Once you’ve done this a few times, it only really
takes a few minutes on each new system, and going forward, it’ll save
you a lot of mucking about with web interfaces. When you take your new
Polycom phone out of the box, plug it into your network, watch it
autoconfigure itself, and then successfully register with your Asterisk
machine, you will know the sort of joy that only geeks can
experience.<sup>[<a id="id4118654" href="#ftn.id4118654">61</a>]</sup></p><p>It’s not really that complicated. Where we think people get
confused is in making sense of the various ways this can be achieved,
because there are a lot of choices.</p><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="dhcp_server"></a>DHCP server</h4></div></div></div><p>Typically, a DHCP<a id="I_indexterm4_tt527" class="indexterm"></a> server is used to configure basic IP parameters for a
device (IP address, default gateway, and DNS), but the DHCP protocol
can actually pass many other parameters. In our case, we want it to
pass some information to the sets that will tell them where to
download their config files from. Here is a sample config from a Linux
DHCP server that will do what is required:</p><p><a id="I_programlisting4_tt528"></a></p><pre class="programlisting">ddns-update-style interim;
ignore client-updates;
subnet 192.168.1.0 netmask 255.255.255.0 {
option routers 192.168.1.1;
option subnet-mask 255.255.255.0;
option domain-name-servers 192.168.1.1;
option ntp-servers pool.ntp.org;
option time-offset -18000;
range dynamic-bootp 192.168.1.128 192.168.1.254;
default-lease-time 21600;
max-lease-time 43200;
}</pre><p>Keep in mind that this assumes that the only things on this
network are devices that belong to the phone system (this setup will
hand out an IP address to any device that requests it). If you have a
more complex environment, you will need to configure the DHCP daemon
to handle the various devices it is serving. For example, you might
want to devise a scope that restricts IP addresses in your voice LAN
to Polycom phones. Since all Polycom IP desk phones have 00:04:f2 as
their OUI (Organizationally Unique Identifier), you might choose to
restrict scope based on that.</p><div class="note" style="margin-left: 0.5in; margin-right: 0.5in;"><h3 class="title">Note</h3><p>In a Microsoft DHCP environment, <a id="I_indexterm4_tt529" class="indexterm"></a>the tftp-server-name is referred to as<a id="I_indexterm4_tt530" class="indexterm"></a> <span class="emphasis"><em>Boot server host name</em></span>. It is
defined under <span class="emphasis"><em>option 66</em></span>.</p></div><p>The DHCP protocol is far more flexible than is often realized,
because in most environments it is not used for complex provisioning
tasks. With a little care and attention, you can devise a DHCP
environment that serves both your voice and data devices and greatly
simplifies administrative workload when adding new devices.</p></div><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="id4118762"></a>FTP server</h4></div></div></div><p>FTP is currently our<a id="I_indexterm4_tt531" class="indexterm"></a> favorite<sup>[<a id="id4118781" href="#ftn.id4118781">62</a>]</sup>way to configure Polycom sets. We would recommend it
over<a id="I_indexterm4_tt532" class="indexterm"></a> TFTP for any set that allows for both. To install it on
your CentOS system, the following command will install<a id="I_indexterm4_tt533" class="indexterm"></a><a id="I_indexterm4_tt534" class="indexterm"></a> VSFTPD, the Very Secure FTP Daemon:</p><p><a id="I_programlisting4_tt535"></a></p><pre class="programlisting"># <strong class="userinput"><code>yum -y install vsftpd</code></strong></pre><p>Then,
in order to lock things down, we need to prevent anonymous logins with
a simple change<a id="I_indexterm4_tt536" class="indexterm"></a> to the <span class="emphasis"><em>vsftpd</em></span> config file,
<code class="filename">/etc/vsftpd/vsftpd.conf</code>:</p><p>
<a id="I_programlisting4_tt537"></a></p><pre class="programlisting"># anonymous_enable=NO</pre><p>
</p><p>Restart the server with <code class="literal">service vsftpd
restart</code>. To ensure that the daemon runs after every reboot,
run <code class="literal">chkconfig vsftpd on</code>.</p><p>Now, we have to create a user account and group for the sets to
use. In this case, we will create an account for the Polycom
sets:</p><p>
<a id="I_programlisting4_tt538"></a></p><pre class="programlisting"># groupadd PlcmSpIp
# useradd PlcmSpIp -g PlcmSpIp -p PlcmSpIp
# passwd PlcmSpIp</pre><p>
</p><p>Set the password to <span><strong class="command">PlcmSpIp</strong></span>
(the default FTP password for Polycom sets). This can be changed, but
will then require manual configuration from each set in order to
advise them of their nonstandard credentials.<sup>[<a id="id4118921" href="#ftn.id4118921">63</a>]</sup></p><p>For added security, let’s make sure the FTP server keeps that
account in a chroot jail:</p><p><a id="I_programlisting4_tt539"></a></p><pre class="programlisting"># echo PlcmSpIp >> /etc/vsftpd/vsftpd.chroot_list</pre><p>That
pretty much does it as far as preparing the operating system to
provide the required services to the phones.</p><p>In the next few sections we have provided instructions for
various popular SIP telephones. Choose the section that applies best
to the phone that you are planning to use (whether a hard- or
soft-phone). You will note that we have given all of these phones the
exact same unique identifier. If you plan on installing more than one
of them, you will need to ensure that they have unique names, and be
sure to update your <code class="filename">sip.conf</code> file
to include those device definitions.</p></div></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4118972"></a>CounterPath’s X-Lite Softphone</h3></div></div></div><p>CounterPath’s X-Lite softphone<a id="ch04_conterpath" class="indexterm"></a><a id="ch04_xlite" class="indexterm"></a> has become very popular with the Asterisk community. It
is simple, functional, easy on the eyes, and—most
importantly—free.</p><p>In this section we will be configuring the X-Lite softphone to
connect to Asterisk. The<a id="I_indexterm4_tt540" class="indexterm"></a> IP address of the phone is 192.168.1.250, and Asterisk is
located at 192.168.1.100. The X-Lite is available for Microsoft Windows,
Mac, and Linux. You can obtain a copy of X-Lite from <a href="http://www.counterpath.com/index.php?menu=download" target="_top">http://www.counterpath.com/index.php?menu=download</a>.</p><p>Now let’s configure our softphone for connecting to our Asterisk
box. To configure X-Lite, click on the Settings button, as circled in
<a href="asterisk-CHP-4-SECT-7.html#asterisk-CHP-4-FIG-2" title="Figure 4.2. X-Lite configuration">Figure 4.2, “X-Lite configuration”</a>.</p><div class="figure"><a id="asterisk-CHP-4-FIG-2"></a><p class="title"><b>Figure 4.2. X-Lite configuration</b></p><div class="mediaobject"><a id="I_mediaobject4_tt541"></a><img src="figs/web/ast2_0402.png" alt="X-Lite configuration" /></div></div><p>Select <span class="emphasis"><em>System Settings</em></span> → <span class="emphasis"><em>SIP
Proxy</em></span> → <span class="emphasis"><em>[Default]</em></span>, which will display
the default configuration for the softphone. Configure the screen as
shown in <a href="asterisk-CHP-4-SECT-7.html#asterisk-CHP-4-FIG-3" title="Figure 4.3. X-Lite user configuration">Figure 4.3, “X-Lite user configuration”</a>.</p><div class="figure"><a id="asterisk-CHP-4-FIG-3"></a><p class="title"><b>Figure 4.3. X-Lite user configuration</b></p><div class="mediaobject"><a id="I_mediaobject4_tt542"></a><img src="figs/web/ast2_0403.png" alt="X-Lite user configuration" /></div></div><p>If you have not already started Asterisk, then start it now (see
<a href="asterisk-CHP-3.html" title="Chapter 3. Installing Asterisk">Chapter 3, <i>Installing Asterisk</i></a> for help installing and starting
Asterisk). If Asterisk is running in the background, you can reconnect
to the CLI by running the following command:</p><p>
<a id="I_programlisting4_tt543"></a></p><pre class="programlisting"># <strong class="userinput"><code>asterisk -rvvv</code></strong></pre><p>
</p><p>You will then be given the Asterisk CLI like so:<a id="I_programlisting4_tt544"></a></p><pre class="programlisting">*CLI></pre><p>If Asterisk was already
running before changing the <code class="filename">sip.conf</code> as instructed in this chapter, then
reload the dialplan and SIP channel with the following two
commands:</p><p><a id="I_programlisting4_tt545"></a></p><pre class="programlisting">*CLI> <strong class="userinput"><code>dialplan reload</code></strong>
*CLI> <strong class="userinput"><code>sip reload</code></strong></pre><p>In your X-Lite
softphone client, close the Settings windows by clicking the BACK button
until the windows are all closed. You should see X-Lite try to register
to Asterisk, and if successful, you will see the following at the
Asterisk CLI:</p><p><a id="I_programlisting4_tt546"></a></p><pre class="programlisting">-- Registered SIP '1000' at 192.168.1.250 port 5061 expires 3600</pre><p>You
can verify the registration status at any time like so:</p><p>
<a id="I_programlisting4_tt547"></a></p><pre class="programlisting">*CLI> <strong class="userinput"><code>sip show peers</code></strong>
Name/username Host Dyn Nat ACL Port Status
1000/1000 192.168.1.250 D N 5061 OK (63 ms)
1 sip peers [1 online , 0 offline]
</pre><p>
</p><p>More detailed stats of the peer can be shown as follows with
<code class="literal">sip show peer 1000</code>:<a id="I_indexterm4_tt548" class="indexterm"></a><a id="I_indexterm4_tt549" class="indexterm"></a></p><p>
<a id="I_programlisting4_tt550"></a></p><pre class="programlisting">*CLI> <strong class="userinput"><code>sip show peer</code></strong> 1000
* Name : 1000
Secret : <Not set>
MD5Secret : <Not set>
Context : phones
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 1032
Insecure : no
Nat : RFC3581
ACL : No
T38 pt UDPTL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 192.168.1.250 Port 5061
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 1000
SIP Options : (none)
Codecs : 0x8000e (gsm|ulaw|alaw|h263)
Codec Order : (none)
Auto-Framing: No
Status : Unmonitored
Useragent : X-Lite release 1105d
Reg. Contact : sip:1000@192.168.1.250:5061
</pre><p>
</p></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4119324"></a>Polycom’s IP 430</h3></div></div></div><p>A lot of folks<a id="ch04_polycom" class="indexterm"></a><a id="ch04_ip430" class="indexterm"></a> say configuring Polycom phones is difficult. From what we
can tell, they base this on one of two reasons: 1) The Polycom web-based
interface is horrible, or 2) the automatic provisioning process is
painful and confusing.</p><p>With respect to item 1, we agree. The web interface on the Polycom
phones has got to be one of the most annoying web interfaces ever
developed for an IP telephone. We don’t use it, and we don’t recommend
it.<sup>[<a id="id4119368" href="#ftn.id4119368">64</a>]</sup></p><p>So that leaves us with some sort of server-based configuration.
Fortunately, in this regard, the Polycom IP phones are superb—so much so
that we can pretty much forgive the web interface. Set configurations
are stored in files on a server, and each set navigates to the server,
downloads the configuration files that are relevant to it, and applies
them to itself.</p><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="id4119384"></a>DHCP server</h4></div></div></div><p>If you cannot<a id="I_indexterm4_tt551" class="indexterm"></a> control your DHCP server, you may have to manually
specify the FTP server information on the phone. This is done by
rebooting the set, pressing the <span><strong class="command">setup</strong></span> button before the set begins the load
process, and specifying the address of the FTP server in the small
boot menu that these phones offer.</p></div><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="id4119416"></a>Protocol to use for downloading</h4></div></div></div><p>The Polycom phones are able to download<a id="I_indexterm4_tt552" class="indexterm"></a> their configuration by one of three protocols: TFTP,
HTTP, and FTP.</p><p>Right off the bat we are going to tell you to avoid<a id="I_indexterm4_tt553" class="indexterm"></a> TFTP. It is not secure, and the set cannot use date
information to determine which versions of various files are the most
current. It works, but there are better ways, and we are not going to
discuss it further.</p><p>Polycom phones can pull their config data using HTTP as well,
but it <a id="I_indexterm4_tt554" class="indexterm"></a>has not proven to be popular, and so we are going to
move on.</p><p>FTP is currently the preferred method of allowing Polycom phones
to obtain their configuration. It works well, is fairly easy to
configure, and is well supported by the community.</p></div><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="id4119475"></a>FTP</h4></div></div></div><p>FTP is <a id="I_indexterm4_tt555" class="indexterm"></a>currently our favorite way to configure Polycom sets. To
install it on your CentOS system, the following command will
<a id="I_indexterm4_tt556" class="indexterm"></a>install VSFTPD, the Very Secure FTP Daemon:</p><p><a id="I_programlisting4_tt557"></a></p><pre class="programlisting"># <strong class="userinput"><code>yum -y install vsftpd</code></strong></pre><p>Then,
in order to lock things down, we need to prevent anonymous logins,
with a simple change to the<a id="I_indexterm4_tt558" class="indexterm"></a><span class="emphasis"><em>vsftpd</em></span> config file, <code class="filename">/etc/vsftpd/vsftpd.conf</code>:</p><p>
<a id="I_programlisting4_tt559"></a></p><pre class="programlisting"># anonymous_enable=NO</pre><p>
</p><p>Restart the server with <code class="literal">service vsftpd
restart</code>. To ensure that the daemon runs after every reboot,
run <code class="literal">chkconfig vsftpd on</code>.</p><p>Now, we have to create a user account and group for the Polycom
sets to use:</p><p>
<a id="I_programlisting4_tt560"></a></p><pre class="programlisting"># groupadd PlcmSpIp
# useradd PlcmSpIp -g PlcmSpIp -p PlcmSpIp
# passwd PlcmSpIp
</pre><p>
</p><p>Set the password to <span><strong class="command">PlcmSpIp</strong></span>
(the default FTP password for Polycom sets). This can be changed, but
will then require manual configuration from each set in order to
advise them of their nonstandard credentials.<sup>[<a id="id4119604" href="#ftn.id4119604">65</a>]</sup></p><p>For added security, let’s make sure the FTP server keeps that
account in a chroot jail:</p><p><a id="I_programlisting4_tt561"></a></p><pre class="programlisting"># echo PlcmSpIp >> /etc/vsftpd/vsftpd.chroot_list</pre><p>That
pretty much does it as far as preparing the operating system to
provide the required services to the phones.</p></div><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="id4119636"></a>The Polycom configuration files</h4></div></div></div><p>While there seem<a id="I_indexterm4_tt562" class="indexterm"></a> to be a lot of different files that are needed to make
a Polycom set work, they are each fairly easy to understand.</p><div class="sect4" lang="en" xml:lang="en"><div class="titlepage"><div><div><h5 class="title"><a id="id4119660"></a>The bootROM</h5></div></div></div><p>This can best be described<a id="I_indexterm4_tt563" class="indexterm"></a> as the BIOS and operating system of the phone.
Perhaps there is a more technical explanation, but why make things
confusing? The bootROM should not need to be updated regularly, but
it is good to keep an eye on the current releases to see if a newer
bootROM has features that will be of benefit in your environment.
This file will be named <code class="filename">bootrom.ld</code>.</p></div><div class="sect4" lang="en" xml:lang="en"><div class="titlepage"><div><div><h5 class="title"><a id="id4119690"></a>The application image</h5></div></div></div><p>Since Polycom sets are capable of supporting other VoIP
protocols (MGCP is supported, for example), the protocol that this
set will employ forms part of the application image that the phone
will download and run. If the image on the set is already correct,
this file is not actually needed on the FTP server; however, it is
common to have this file available to ensure that the most recent
version of the protocol is available for the sets to download. You
will sometimes receive phones that are not running the latest
version, so having the most current image will ensure that all sets
are up-to-date.</p></div><div class="sect4" lang="en" xml:lang="en"><div class="titlepage"><div><div><h5 class="title"><a id="id4119705"></a>The sip.cfg file</h5></div></div></div><p>There is normally only one <a id="I_indexterm4_tt564" class="indexterm"></a>version of this file on a system, but it can be named
anything you want, and there can be as many different versions of
this file as are needed. For example, if you had an office where
there were two different languages in use, some users might prefer
French on their set, and others English. In that case, you’d create
a <code class="filename">french.sip.conf</code> file and an
<code class="filename">english.sip.conf</code> file to handle
each case. Name this file as you see fit, but pick a name that makes
sense so that future administrators have a chance to make sense of
your design choices.</p></div><div class="sect4" lang="en" xml:lang="en"><div class="titlepage"><div><div><h5 class="title"><a id="id4119746"></a>The master config file for each phone</h5></div></div></div><p>This file is very simple and small. It is named to match the
<a id="I_indexterm4_tt565" class="indexterm"></a>MAC address of each phone (so each set will need its
own copy of this file) and tells the set what other files it needs
to download in order to configure itself. This is the first config
file each set will read. In this file will be a reference to the
application image this set will use (currently named
<span class="emphasis"><em>sip.ld</em></span>), as well as the names of the XML files
that have the parameters for this phone (the
<span class="emphasis"><em>.cfg</em></span> files). A master config file for a set
might look something like this:</p><a id="I_programlisting4_tt566"></a><pre class="programlisting">'<?xml version="1.0" standalone="yes"?>'
'<!-- Default Master SIP Configuration File-->'
'<!-- Edit and rename this file to <Ethernet-address>.cfg for each phone.-->'
'<!-- $Revision: 1.14 $ $Date: 2005/07/27 18:43:30 $ -->'
'<APPLICATION APP_FILE_PATH="sip.ld"
CONFIG_FILES="phone1.cfg, sip.cfg"
MISC_FILES=""
LOG_FILE_DIRECTORY=""
OVERRIDES_DIRECTORY=""
CONTACTS_DIRECTORY=""
/>'
</pre><p>Note the name of the application file that we want this set to
use, and the config files that it will be trying to find and
apply.</p></div><div class="sect4" lang="en" xml:lang="en"><div class="titlepage"><div><div><h5 class="title"><a id="id4119800"></a>The set-specific config file</h5></div></div></div><p>We recommend giving the <code class="filename">phone1.cfg</code> files<a id="I_indexterm4_tt567" class="indexterm"></a> names that make sense. For example, <code class="filename">SET<xxx>.cfg</code> (such as <code class="filename">SET201.cfg</code>) to match the extension number
of the phone, or <code class="filename">FLOOR4CUBE23.cfg</code>, or maybe <code class="filename">BOB_SMITHS_IP430_SET.cfg</code>, or whatever
seems best to you. What’s the best way to name them? We’re going to
answer that question by asking a question. Let’s say you have 100 of
these phones. When you list the contents of the <code class="filename">/home/PlcmSpIp</code> folder, how do you want
the 100 config files for the sets to appear?</p></div><div class="sect4" lang="en" xml:lang="en"><div class="titlepage"><div><div><h5 class="title"><a id="id4119863"></a>Gotchas</h5></div></div></div><p>Settings that are configured directly on the telephone will be
stored on the filesystem of the set, and may take precedence over
parameters passed in config files. If you are having any problems
applying changes to a set, try reformatting the phone. This will
force the set to accept the parameters contained in the config
files.<a id="I_indexterm4_tt568" class="indexterm"></a><a id="I_indexterm4_tt569" class="indexterm"></a></p></div></div></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="cisco_7960"></a>Cisco 7960 Telephone</h3></div></div></div><p>The venerable<a id="ch04_cisco7960" class="indexterm"></a><a id="ch04_7960" class="indexterm"></a><a id="ch04_c7960" class="indexterm"></a> old C7960 is now a part of VoIP history. One of the first
SIP telephones that could actually be taken seriously, the only real
complaint one can have about this phone is the price: they are the
Cadillac of SIP phones (meaning that they have all the bells and
whistles but are tough to justify at the price, and are a little out of
date <span class="keep-together">sometimes).</span></p><p>If you can get one of these, you are getting an excellent SIP
telephone. If you buy one new, be prepared to pay.</p><p>One of the ways this phone is out of date is the lack of remote
provisioning from anything other than<a id="I_indexterm4_tt570" class="indexterm"></a> TFTP. TFTP has lost favor with networking professionals
due to the lack of authentication and encryption, but since it is the
only method of remotely provisioning the phone, we are going to have to
use the tftp-server daemon. We can install tftp-server with the
following command:</p><p><a id="I_programlisting4_tt571"></a></p><pre class="programlisting"># <strong class="userinput"><code>yum install -y tftp-server</code></strong></pre><p>Once
installed, we need to enable the server by modifying the
<span class="emphasis"><em>/etc/xinetd.d/tftp</em></span> file. To enable the TFTP server,
change the <code class="literal">disable=yes</code> line to
<code class="literal">disable=no</code>.</p><p><a id="I_programlisting4_tt572"></a></p><pre class="programlisting">service tftp
{
socket_type = dgram
protocol = udp
wait = yes
user = root
server = /usr/sbin/in.tftpd
server_args = -s /tftpboot
<strong class="userinput"><code> disable = no</code></strong>
per_source = 11
cps = 100 2
flags = IPv4
}
</pre><p>Then start the TFTP server by running:</p><p><a id="I_programlisting4_tt573"></a></p><pre class="programlisting"># <strong class="userinput"><code>service xinetd restart</code></strong></pre><p>We
can verify the server is running with the following command:</p><p><a id="I_programlisting4_tt574"></a></p><pre class="programlisting"># <strong class="userinput"><code>chkconfig --list | grep tftp</code></strong>
tftp: on</pre><p>As long as <code class="literal">tftp:
on</code> was returned, the server is up and running.</p><div class="note" style="margin-left: 0.5in; margin-right: 0.5in;"><h3 class="title">Note</h3><p>Cisco phones by default are loaded with their own
communication protocol known<a id="I_indexterm4_tt575" class="indexterm"></a><a id="I_indexterm4_tt576" class="indexterm"></a> as SCCP (or Skinny). We will be showing you how to
configure the phone, but due to the proprietary nature of Cisco and
its phones, you will need to obtain the SIP firmware from your
distributor. Also, there are both <span class="emphasis"><em>chan_sccp</em></span> and
<span class="emphasis"><em>chan_skinny</em></span> modules for Asterisk, but they are
beyond the scope of this book.</p></div><p>We will be registering our Cisco phone to the SIP friend we
configured in <a href="asterisk-CHP-4-SECT-6.html#asterisk-CHP-4-SECT-6.1" title="Zaptel Hardware Configuration">the section called “Zaptel Hardware Configuration”</a>.” The following
configuration file should be saved into a file taking the format of
<code class="filename">SIP<mac>.cnf</code>, where <code class="filename"><mac></code> represents the MAC address of the
telephone device you are configuring. Place this file into the <code class="filename">/tftpboot/</code> directory on your server:</p><p><a id="I_programlisting4_tt577"></a></p><pre class="programlisting"># Line 1 Configuration
line1_name: "1000"
line1_authname: "1000"
line1_shortname: "Jimmy Carter"
line1_password: ""
line1_displayname: ""
# The phone label, displayed in the upper-righthand corner of the phone
phone_label: "aristotle" ; Has no effect on SIP messaging
# Phone password used for console or telnet access, limited to 31 characters
phone_password: "cisco"</pre><p>Then configure the address to register
in the <code class="filename">SIPDefault.cnf</code><a id="I_indexterm4_tt578" class="indexterm"></a>file, also placed in the <span class="keep-together"><code class="filename">/tftpboot/</code></span> directory of your server.
<code class="literal">proxy1_address</code> will contain the IP address of your
Asterisk server of where the phone should register for line 1. The
<code class="literal">image_version</code> contains the version of the
<span class="emphasis"><em>.loads</em></span> and <span class="emphasis"><em>.sb2</em></span> files the
phone will load into memory.</p><p><a id="I_programlisting4_tt579"></a></p><pre class="programlisting">image_version: P0S3-08-4-00
proxy1_address: 192.168.1.100</pre><p>We need one additional
file<a id="I_indexterm4_tt580" class="indexterm"></a> called <span class="emphasis"><em>OS79XX.TXT</em></span>. This file
contains only a single line―the <span class="emphasis"><em>.bin</em></span> and
<span class="emphasis"><em>.sbn</em></span> file version to load into memory:</p><p><a id="I_programlisting4_tt581"></a></p><pre class="programlisting">P003-08-4-00</pre><p>In
order for our Cisco 7960 to use these files, we need to tell the phone
where to pull its configuration from. If using the DHCP server from your
Linux server, you can modify the <code class="filename">/etc/dhcpd.conf</code> file in order to tell the
phone where to pull its configuration from by adding the line:</p><a id="I_programlisting4_tt582"></a><pre class="programlisting">option tftp-server-name "192.168.1.100";</pre><p>which contains the IP address of the server hosting the TFTP
server (assuming of course the TFTP server is configured at that
address. This is the address we’ve been using for our Asterisk server,
and we again assume you’ve installed the TFTP server on the same box as
Asterisk). See <a href="asterisk-CHP-4-SECT-7.html#dhcp_server" title="DHCP server">the section called “DHCP server”</a>” for more information
about configuring the DHCP server:</p><p>
<a id="I_programlisting4_tt583"></a></p><pre class="programlisting">ddns-update-style interim;
ignore client-updates;
subnet 192.168.1.0 netmask 255.255.255.0 {
option routers 192.168.1.1;
option subnet-mask 255.255.255.0;
option domain-name-servers 192.168.1.1;
<strong class="userinput"><code> option tftp-server-name "192.168.1.100";</code></strong>
option ntp-servers pool.ntp.org;
option time-offset -18000;
range dynamic-bootp 192.168.1.128 192.168.1.254;
default-lease-time 21600;
max-lease-time 43200;
}</pre><p>
</p><div class="tip" style="margin-left: 0.5in; margin-right: 0.5in;"><h3 class="title">Tip</h3><p>Alternatively, you can configure from the phone itself to
manually use an alternative TFTP server than that given by the DHCP
server. To do so, press the <span class="emphasis"><em>settings</em></span> button,
(on the G version of the Cisco phones, this looks like a square with
a check mark inside of it; G means Global). You will then need to
unlock the settings by pressing the 9 key. The default password is
<span class="emphasis"><em>cisco</em></span>. Once the phone is unlocked, press the 3
key on the dialpad to enter the Network Configuration. Scroll down
to option 32 and set the Alternate TFTP to YES. Then scroll up to
option 7 and enter the IP address of the TFTP server you wish to
boot from. Accept the settings and back out of the menu until the
phone reboots itself. You can also use the
<span class="emphasis"><em>*-6-settings</em></span> three finger salute to reboot your
phone at any time.</p></div><p>You can watch the phone pull its configuration from the TFTP
server by using <code class="literal">tshark</code> (<code class="literal">yum install ethereal</code>). Filter on port 69 using
the following command:</p><p><a id="I_programlisting4_tt584"></a></p><pre class="programlisting"># <strong class="userinput"><code>tshark port 69</code></strong></pre><p>You
should then be able to watch the network traffic from the phone
requesting data from your TFTP server.</p><p>If all goes well, then you should see your phone registered to
<a id="I_indexterm4_tt585" class="indexterm"></a><a id="I_indexterm4_tt586" class="indexterm"></a><a id="I_indexterm4_tt587" class="indexterm"></a>Asterisk!</p></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4120413"></a>Linksys SPA-942</h3></div></div></div><p>Ever since they<a id="ch04_linksys942" class="indexterm"></a><a id="ch04_942" class="indexterm"></a> purchased<a id="I_indexterm4_tt588" class="indexterm"></a> Sipura Technologies, Linksys has been producing a line of
economical VoIP telephones and<a id="I_indexterm4_tt589" class="indexterm"></a> <a id="I_indexterm4_tt590" class="indexterm"></a>ATAs (Analog Terminal Adaptors). Linksys has been stealing
a lot of business from Cisco. If you have read<a id="I_indexterm4_tt591" class="indexterm"></a> Clayton M. Christensen’s <span class="emphasis"><em>The Innovator’s
Dilemma</em></span> (HarperCollins), <a id="I_indexterm4_tt592" class="indexterm"></a>it becomes easier to understand Cisco’s strategy with
respect to Linksys.</p><p>Linksys (and Sipura) products are well regarded for their
excellent quality, especially relative to their price, but they are also
famous for being painfully difficult to configure. This is mostly
because their configuration GUI offers hundreds of configurable
parameters.</p><p>We don’t care about that. Here’s what you need to know to get an
SPA-942 working with your Asterisk system (and, we hope, most other
Linksys VoIP devices as well).</p><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="id4120521"></a>Logging in to the phone</h4></div></div></div><p>The first thing you need to do is get the IP address of the
phone so you can log in to the GUI interface. From the phone itself,
select the icon that looks like a piece of paper with a dog-eared
corner (right below the envelope icon). This is the Settings button,
and is shown in <a href="asterisk-CHP-4-SECT-7.html#asterisk-CHP-4-FIG-4" title="Figure 4.4. SPA-942 keypad">Figure 4.4, “SPA-942 keypad”</a>.</p><div class="figure"><a id="asterisk-CHP-4-FIG-4"></a><p class="title"><b>Figure 4.4. SPA-942 keypad</b></p><div class="mediaobject"><a id="I_mediaobject4_tt593"></a><img src="figs/web/ast2_0404.png" alt="SPA-942 keypad" /></div></div><p>To get the IP address of the phone, press the Settings button,
followed by 9 (or use directional pad and scroll down to Network).
Then press the <span class="emphasis"><em>select</em></span> button (there is a row of 4
buttons under the LCD screen—<span class="emphasis"><em>select</em></span> is the
leftmost button). The second field should show you the IP address of
the phone.</p><p>Now open your browser, enter the IP address into the address
bar, hit Enter, and you will be at the Info screen of the
phone.</p></div><div class="sect3" lang="en" xml:lang="en"><div class="titlepage"><div><div><h4 class="title"><a id="id4120601"></a>Registering your phone to Asterisk</h4></div></div></div><p>Select the Admin Login link in the upper-right corner of the
screen. Once you’ve done this, you will be given several new tabs,
such as Regional, Phone, Ext 1, Ext 2, and User.</p><p>Select the Ext 1 tab which will set up our first line. Then make
the following menu selections:</p><div class="orderedlist"><ol type="1"><li><p>General → Line Enable → yes</p></li><li><p>NAT Settings → NAT Mapping Enable → no</p></li><li><p>NAT Settings → NAT Keep Alive Enable → no</p></li><li><p>Proxy and Registration → Proxy → enter the IP address of
Asterisk <span class="keep-together">(e.g.,
192.168.1.100)</span></p></li><li><p>Proxy and Registration → Register → yes</p></li><li><p>Proxy and Registration → Make Call Without Reg → no</p></li><li><p>Proxy and Registration → Ans Call Without Reg → no</p></li><li><p>Subscriber Information → Display Name → Caller ID
information</p></li><li><p>Subscriber Information → User ID → 1000</p></li><li><p>Subscriber Information → Password → (leave blank if you’re
using the simple configuration from earlier in this
chapter)</p></li><li><p>Subscriber Information → Use Auth ID → yes</p></li><li><p>Subscriber Information → Auth ID → 1000</p></li><li><p>Audio Configuration → Preferred Codec → G711u</p></li><li><p>Audio Configuration → Use Pref Codec Only → no</p></li><li><p>Audio Configuration → Silence Supp Enable → no</p></li><li><p>Audio Configuration → DTMF Tx Method → Auto</p></li><li><p>Submit All Changes</p></li></ol></div><p>And that’s it! Your phone should be registered to Asterisk now.
You’ll know this because the first lighted line button beside the LCD
screen will change from orange to green.<a id="I_indexterm4_tt594" class="indexterm"></a><a id="I_indexterm4_tt595" class="indexterm"></a></p></div></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4120735"></a>Configuring the Dialplan for Testing</h3></div></div></div><p>In order to allow our phone to <a id="I_indexterm4_tt596" class="indexterm"></a>call other phones (or, if a multiline phone, to call
itself), we need to modify the <span class="emphasis"><em>extensions.conf</em></span>
file. Building on what we had in <a href="setting_up_dialplan.html" title="Setting Up the Dialplan for Some Test Calls">the section called “Setting Up the Dialplan for Some Test Calls”</a>,” add the following parts to the
<code class="literal">[internal]</code> context:</p><p><a id="I_programlisting4_tt597"></a></p><pre class="programlisting">exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()</pre><p>If you have two phones, or
multiple lines configured, you can duplicate the previous configuration
and change the 1000 to the other extension <a id="I_indexterm4_tt598" class="indexterm"></a>name.</p></div><div class="footnotes"><br /><hr width="100" align="left" /><div class="footnote"><p><sup>[<a id="ftn.id4117935" href="#id4117935">57</a>] </sup>RFC 3261<a id="I_indexterm4_tt507" class="indexterm"></a> is available at <a href="http://www.ietf.org/rfc/rfc3261.txt" target="_top">http://www.ietf.org/rfc/rfc3261.txt</a>.
While the document is fairly large, we strongly encourage anyone who
wishes to become an Asterisk professional to read at least the first
100 or so pages of this document and to understand how calls are set
up, as this knowledge will be imperative when you’re looking at a SIP
trace (<code class="literal">sip debug</code> from the Asterisk console) trying
to determine why your calls are not getting set up correctly.</p></div><div class="footnote"><p><sup>[<a id="ftn.id4118046" href="#id4118046">58</a>] </sup>An excellent open source SIP proxy is<a id="I_indexterm4_tt512" class="indexterm"></a><a id="I_indexterm4_tt513" class="indexterm"></a> OpenSER, available at <a href="http://www.openser.org" target="_top">http://www.openser.org</a>.</p></div><div class="footnote"><p><sup>[<a id="ftn.id4118104" href="#id4118104">59</a>] </sup>Yes, there are ways to making the media flow directly between
the phones once the call is set up. This is done in the
<span class="emphasis"><em>sip.conf</em></span> file using either
<code class="literal">directrtpsetup=yes </code>(an experimental option allowing
the media to be redirected in the initial call setup) or
<code class="literal">canreinvite=yes</code> (where media initially goes through
Asterisk until a re-INVITE happens, at which point the media can be
sent directly between the phones).</p></div><div class="footnote"><p><sup>[<a id="ftn.id4118250" href="#id4118250">60</a>] </sup>The maximum length of a username is 255 characters.</p></div><div class="footnote"><p><sup>[<a id="ftn.id4118654" href="#id4118654">61</a>] </sup>Typically, it’s at 4 A.M. on the morning of a critical 8 A.M.
meeting, after having worked all weekend. Red Bull is probably the
most popular drink of the Asterisk developers. Dr. Pepper would be a
close second. Red Bull, anyone?</p></div><div class="footnote"><p><sup>[<a id="ftn.id4118781" href="#id4118781">62</a>] </sup>FTP is preferred over TFTP due to the ability of a Polycom
phone to see timestamps on FTP files. This allows the phone to
avoid redownloading configuration files and firmware updates that
it already has—thus shortening boot time.</p></div><div class="footnote"><p><sup>[<a id="ftn.id4118921" href="#id4118921">63</a>] </sup>You can get into assigning complex and unguessable passwords
for the phones to use, but unless you are going to input the
passwords into each phone manually, you’ll have to pass them their
FTP username and password from the DHCP server. Any device that
can get on the voice network can get the same information from the
DHCP server. We’re not telling you to ignore security; just don’t
assume that creating separate passwords for each phone is going to
improve security.</p></div><div class="footnote"><p><sup>[<a id="ftn.id4119368" href="#id4119368">64</a>] </sup>Actually, it does serve one useful purpose, which is to allow
you to log on to a set via a browser and query its
configuration.</p></div><div class="footnote"><p><sup>[<a id="ftn.id4119604" href="#id4119604">65</a>] </sup>You can get into assigning complex and unguessable passwords
for the phones to use, but unless you are going to input the
passwords into each phone manually, you’ll have to pass them their
FTP user name and password from the DHCP server. Any device that
can get on the voice network can get the same information from the
DHCP server. We’re not telling you to ignore security, just don’t
assume that creating separate passwords for each phone is going to
improve security.</p></div></div></div><div class="navfooter"><hr /><table width="100%" summary="Navigation footer"><tr><td width="40%" align="left"><a accesskey="p" href="asterisk-CHP-4-SECT-6.html">Prev</a> </td><td width="20%" align="center"><a accesskey="u" href="asterisk-CHP-4.html">Up</a></td><td width="40%" align="right"> <a accesskey="n" href="I_sect14_tt599.html">Next</a></td></tr><tr><td width="40%" align="left" valign="top">Configuring an FXS Channel for an Analog Telephone </td><td width="20%" align="center"><a accesskey="h" href="index.html">Home</a></td><td width="40%" align="right" valign="top"> Connecting to a SIP Service Provider</td></tr></table></div><div xmlns="" id="svn-footer"><hr /><p>You are reading <em>Asterisk: The Future of Telephony</em> (2nd Edition for Asterisk 1.4), by Jim van Meggelen, Jared Smith, and Leif Madsen.<br />
This work is licensed under the <a href="http://creativecommons.org/licenses/by-nc-nd/3.0/">Creative Commons Attribution-Noncommercial-No Derivative Works License v3.0</a>.<br />
To submit comments, corrections, or other contributions to the text, please visit <a href="http://oreilly.com/catalog/9780596510480/">http://www.oreilly.com/</a>.</p></div></body></html>
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