SIP.edu CSU Deployment Notes

3.2.4. Modifying /etc/asterisk/extensions.conf

This is where incoming calls sent by SER are handled. We set the caller ID name to the caller name provided in the SIP header (contained in the ${CALLERIDNAME} variable), and the number to 000-000-0001, (which is an invalid phone number). The content of the file is:

[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
[proxy]
exten => _970NXXXXXX,1,SetCallerID,"sip:${CALLERIDNAME}<0000000001>"
exten => _970NXXXXXX,2,Dial(Zap/g1/${EXTEN:3})
exten => _970NXXXXXX,3,Hangup
[default]
include => proxy

3.3. Configuring SER

The SER package installs a '/usr/sbin/serctl' script. A line containing the SIP domain must be added to the top of this file:

SIP_DOMAIN="colostate.edu"

We currently do not offer registration for user agents. SER's only purpose is to forward all SIP requests to Asterisk. We therefore do not need authentication or database support for now (we do plan to offer user-agent registration in the future, though). We use an LDAP lookup script based on the example provided by Columbia.edu to obtain phone numbers from the first.last@colostate.edu email-like SIP URIs.

Our /etc/ser/ser.cfg file looks like this:

check_via=no    # (cmd. line: -v)
dns=no           # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
listen=AAA.BBB.CCC.DDD		# this is where SER listens (port 5060)
alias="colostate.edu"		# this is the default domain we serve
alias="AAA.BBB.CCC.DDD"		# (ACK and BYE) URIs will also have Asterisk's IP:5065 !
fifo="/tmp/ser_fifo"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/exec.so"
modparam("usrloc", "db_mode",   0)
modparam("rr", "enable_full_lr", 1)
route{
        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                break;
        };
        if ( msg:len > max_len ) {
                sl_send_reply("513", "Message too big");
                break;
        };
        record_route(); 
        if (loose_route()) {
                t_relay();
                break;
        };

	################################################################
	# until here, everything is pretty much the standard boilerplate
	# material that is shipped in the default SER config file.
	################################################################

        # reject REGISTER attempts for now
        if (method=="REGISTER") {
                sl_send_reply( "503", "Registration Unavailable" );
                break;
        };

        # resolve alphanumeric names in the URI
        if (uri=~"sip:[a-zA-Z\.]*@colostate\.edu") {
                exec_dset( "/etc/ser/siplookup" );
        };

        # forward all calls to sip:970XXXYYYY@colostate.edu to Asterisk
        # ACK and BYE requests sometimes have URI @AAA.BBB.CCC.DDD:5065 (i.e., inteneded for Asterisk!)
        # uri==myself would match anything @colostate.edu or @AAA.BBB.CCC.DDD
        # we just want specific phone numbers @colostate.edu to match
        if (uri=~"sip:970[0-9]{7}@(colostate\.edu)|(AAA\.BBB\.CCC\.DDD)" ) {
                forward( localhost, 5065 );
                break;
        };

        # send not found error for any other URI
        sl_send_reply( "404", "Not Found" );

}

The siplookup script is based on Columbia.edu's example. We have a few hard-coded email-phone translations first (e.g., my desktop phone is listed in LDAP, but I want my cellphone to ring when a SIP.edu call comes in). If no hard-coded exceptions are found, the LDAP directory is used to figure out the phone number. Here's our siplookup script:

#!/bin/bash

# convert   sip:@colostate.edu   into   sip:970XXXYYYY@colostate.edu
#
# Gabriel L. Somlo, 2004-12-07
#

LDAP_SERV="ldapserver.colostate.edu"
LDAP_BASE="dc=colostate,dc=edu"

if [ -z "${1}" ]; then
  echo
  echo "   Usage:  $0 sip:@colostate.edu"
  echo
  echo "   Returns the corresponding numerical SIP URI:"
  echo
  echo "       sip:970XXXYYYY@colostate.edu"
  echo
  exit 1
fi

EMAIL=$(echo ${1} | cut -d: -f2)

# look up hardcoded special cases first:
PHONEINFO=$(grep -i "^${EMAIL} " << EOT
Joe.Bloggs@colostate.edu 9705671017
EOT)
PHONE=${PHONEINFO#* }

# search LDAP directory if special-case hardcoded number not found:
if [ -z "${PHONE}" ]; then
  PHONE=$(ldapsearch -LLL -x -h ${LDAP_SERV} -b ${LDAP_BASE} mail=${EMAIL} telephonenumber | grep -i telephonenumber | cut -d' ' -f2 | tr -d '-')
fi

# print out original unmodified URI if nothing found, or @colostate.edu
if [ -z "${PHONE}" ]; then
  echo "${1}"
else
  echo "sip:${PHONE}@colostate.edu"
fi

4. Information for Callers

This information is intended for those who will call SIP.edu users. In essence, if you are one of the users reachable via SIP.edu, this is what you need to show to those who might want to call you.

Currently, this information is only applicable for callers with a valid public IP address. There are workarounds for when the caller is behind a NAT gateway, but that scenario is not (yet) being addressed in this document.

Essentially, the caller will ned a SIP-capable user agent (i.e., "phone" -- either software or hardware, see the Cookbook for details). This document covers the use of X-Lite for Windows and the Mac, and KPhone for Linux. These can be downloaded at the following locations:

• X-Lite: http://www.xten.com/index.php?menu=product&smenu=xlite
• KPhone: http://www.wirlab.net/kphone/
  (a Fedora Core 2 source RPM available at http://www.cs.colostate.edu/~somlo/rpms/kphone-4.1.0-0.i386.rpm)

Registration with a VoIP service provider may be available to the caller, in which case that provider's instructions should be followed. These instructions are intended for standalone callers who do not have any sort of account with any kind of VoIP service provider.

4.1. Configuring X-Lite as a Standalone Client

After downloading X-Lite, the first time the application is started, it will run the user through its "Audio Tuning Wizard" (i.e., it will optimize the audio and microphone settings). The steps are:

1. set the volume to a comfortable level while a recording is being played
2. user talks into the microphone and adjusts gain until playback of the user's recorded voice is satisfactory
3. calibrate the microphone by recording a period of silence to compare against the previous voice recording.

Once the audio setup is completed, the Menu window will open automatically, offering to configure the "SIP Proxy settings". The window looks like this:

The following options must be configured:

• Enabled: Yes
• Display Name: This is what will be displayed on the called party's CallerID; e.g., your first and last name
• Username: This could be your account name on the machine you're currently logged on. You can accept calls if callers dial "sip:<username>@<your-IP-address>"
• Authorization User: leave blank
• Password: leave blank (four *'s will be printed, that's OK)
• Domain/Realm: This could be the fully qualified DNS name of your machine, or just the domain name of the network you're on, or you could simply make up something if all you want is to make outbound calls.
• SIP Proxy: leave blank
• Out Bound Proxy: leave blank
• Use Outbound Proxy: Never
• Send Internal IP: Default (default setting, leave alone)
• Register: Never
• Voicemail SIP URL: leave blank
• Forward SIP URL: leave blank
• Use Voicemail: Forward to Voicemail (default setting, leave alone)
• Direct Dial IP: No (default setting, leave alone)
• Dial Prefix: leave blank
• Provider Website (if applicable): leave blank
• Update Settings (if applicable): leave blank
• Reset: Do not hit enter on this one :)

Hit 'BACK', then close the menu window. X-lite should now tell you that you're "logged in" and to enter a "phone number":

The "phone number" should be a SIP URI (in CSU's case, this looks exactly like the e-mail address listed in the LDAP directory, e.g., <first>.<last>@colostate.edu).

If you missed the original menu configuration window, you can always revisit by clicking the menu button on X-lite. Navigate to 'System Settings', then 'SIP Proxy', then 'Default:'. This will bring back the interface described above, and allow you to make the required setting changes:

4.2. Configuring KPhone as a Standalone Client

When KPhone first starts, it brings up the "Identity" configuration menu, which looks like this:

The following options must be configured:

• Full Name: This is what will be displayed on the called party's CallerID; e.g., your first and last name
• User Part of SIP URL: This could be your account name on the machine you're currently logged on.
• Host Part of SIP URL: This could be the fully qualified DNS name of your machine, or just the domain name of the network you're on, or you could just make something up if all you want is to make outbound calls
• Outbound Proxy (optional): leave blank
• Authentication Username (optional): leave blank
• q-value between 0.0-1.0 (optional): leave blank
• Auto Register: unchecked

You are now ready to dial your party's SIP URI. Click on the "ear" symbol to place the call. To reconfigure your identity, select "File" and then "Identity".The main KPhone window looks like this: