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<channel>
	<title>Linux and Unix technical recipes &#187; Asterisk</title>
	<atom:link href="http://www.howtonix.com/category/asterisk/feed/" rel="self" type="application/rss+xml" />
	<link>http://www.howtonix.com</link>
	<description>Linux Tutorial and Guide</description>
	<lastBuildDate>Thu, 30 Sep 2010 23:41:09 +0000</lastBuildDate>
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		<item>
		<title>Block DDoS attack on SIP Server</title>
		<link>http://www.howtonix.com/block-ddos-attack-sip-server/</link>
		<comments>http://www.howtonix.com/block-ddos-attack-sip-server/#comments</comments>
		<pubDate>Wed, 16 Jun 2010 21:29:45 +0000</pubDate>
		<dc:creator>emran</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[FreeSWITCH]]></category>
		<category><![CDATA[Kamailio]]></category>
		<category><![CDATA[OpenSIPS]]></category>
		<category><![CDATA[PJSIP]]></category>
		<category><![CDATA[Yate]]></category>
		<category><![CDATA[DDoS Attach on SIP Server]]></category>

		<guid isPermaLink="false">http://www.howtonix.com/?p=150</guid>
		<description><![CDATA[google_ad_client = "pub-4492234592142803"; /* 468x60, created 9/1/10 howtonix */ google_ad_slot = "5691725759"; google_ad_width = 468; google_ad_height = 60; Sometimes SIP Server attacked by hacker with huge number of SIP Registration.Which make mad the SIP Server.Full service can be get down. It can be blocked easily by IPTables. IPTables will work as Session Border Controller(SBC) for [...]]]></description>
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<p>Sometimes SIP Server attacked by hacker with huge number of  SIP Registration.Which make mad the SIP Server.Full service can be get down. It can be blocked easily by IPTables. IPTables will work as Session Border Controller(SBC) for SIP Server. This script will work only with SIP.it will not work for TLS or SIPS.</p>
<blockquote><p>
#!/bin/sh<br />
#<br />
# INVITE rate, per host.  Remember a successful (authenticated) call requires 2 INVITEs-<br />
# Initial INVITE, 407 auth required (w/ nonce), INVITE with nonce and authentication.<br />
IRATE=4/minute</p>
<p># REGISTER rate, per host.<br />
RRATE=2/minute</p>
<p># All other SIP methods rate, per host.  Be careful with SUBSCRIBEs, OPTIONS, CANCELs, etc.<br />
ORATE=10/minute</p>
<p># Methods for this script to ignore.  These SIP methods are always allowed.<br />
IGMETH=&#8221;OPTIONS&#8221;</p>
<p># Burst<br />
BURST=1</p>
<p># Interface(s) to protect on INPUT. Seperate multiple interfaces with spaces.<br />
# This will protect SIP services on THIS HOST.<br />
IFACE=&#8221;eth0&#8243;</p>
<p># Reject/drop action &#8211; usually something like DROP or REJECT.<br />
# Use ACCEPT to use this script to not filter traffic but still collect statistics.<br />
DACTION=DROP</p>
<p># Protocol(s) to filter &#8211; can be either tcp or udp or both. Seperate multiples with spaces.<br />
PROTOCOLS=&#8221;udp tcp&#8221;</p>
<p># Enable logging.<br />
#LOG=YES</p>
<p># Block tel: URIs completely?<br />
# P.S. &#8211; tel: sucks!<br />
BLOCKTEL=yes</p>
<p># Interface(s) to protect on FORWARD. Seperate multiple interfaces with spaces.<br />
# The same hashtable will protect the entire network from the same host(s).<br />
# Destination IP is NOT taken into consideration.<br />
# This will protect any SIP services running on the network that uses this machine<br />
# as a router (as long as you get the interfaces right).<br />
#FIFACE=&#8221;eth0&#8243;</p>
<p># Location of iptables binary.<br />
IPTABLES=`which iptables`</p>
<p># Search packet to this location. A larger offset looks further into the packet<br />
# and takes more time but could catch more attacks (and false alarms).<br />
# Remember, the method to match on is always in the beginning of the packet.<br />
OFFSET=65</p>
<p># SIP port<br />
SPORT=5060</p>
<p>if [ ! "$1" ]<br />
then<br />
echo &#8220;SIP DoS/DDoS mitigation script for iptables<br />
See top of script for configuration</p>
<p>Usage:<br />
$0 [start|stop|status]&#8221;<br />
exit 1<br />
fi</p>
<p>if [ "$1" = "status" ]<br />
then<br />
$IPTABLES -L -v -n<br />
exit<br />
fi</p>
<p># Setup iptables<br />
$IPTABLES -F sipdos 2&gt; /dev/null<br />
$IPTABLES -X sipdos 2&gt; /dev/null<br />
$IPTABLES -N sipdos 2&gt; /dev/null</p>
<p>if [ "$1" = "stop" ]<br />
then<br />
echo &#8220;Clearing iptables rules&#8230;&#8221;<br />
if [ "$FIFACE" ]<br />
then<br />
$IPTABLES -F FORWARD 2&gt; /dev/null<br />
fi<br />
$IPTABLES -F INPUT 2&gt; /dev/null<br />
exit<br />
fi</p>
<p># Send the right traffic through our chain<br />
for i in $IFACE<br />
do<br />
for l in $PROTOCOLS<br />
do<br />
$IPTABLES -A INPUT -i $i -m $l -p $l &#8211;dport $SPORT -j sipdos<br />
done<br />
done</p>
<p># Send the right forwarded traffic through our chain<br />
if [ "$FIFACE" ]<br />
then<br />
for j in $FIFACE<br />
do<br />
for l in $PROTOCOLS<br />
do<br />
$IPTABLES -A FORWARD -i $j -m $l -p $l &#8211;dport $SPORT -j sipdos<br />
done<br />
done<br />
fi</p>
<p># &#8220;Handle&#8221; tel: URIs<br />
if [ "$BLOCKTEL" ]<br />
then<br />
$IPTABLES -A sipdos -m string &#8211;string &#8220;tel:&#8221; &#8211;algo bm &#8211;to $OFFSET -j $DACTION<br />
fi</p>
<p># Ignore certain (configured) methods<br />
if [ "$IGMETH" ]<br />
then<br />
for k in $IGMETH<br />
do<br />
$IPTABLES -A sipdos -m string &#8211;string &#8220;$k sip:&#8221; &#8211;algo bm &#8211;to $OFFSET -j ACCEPT<br />
done<br />
fi</p>
<p># Finally set some limits&#8230;</p>
<p># INVITE limit<br />
$IPTABLES -A sipdos -m string &#8211;string &#8220;INVITE sip:&#8221; &#8211;algo bm &#8211;to $OFFSET \<br />
-m hashlimit &#8211;hashlimit $IRATE &#8211;hashlimit-burst $BURST \<br />
&#8211;hashlimit-mode srcip,dstport &#8211;hashlimit-name sip_i_limit -j ACCEPT</p>
<p># REGISTER limit<br />
$IPTABLES -A sipdos -m string &#8211;string &#8220;REGISTER sip:&#8221; &#8211;algo bm &#8211;to $OFFSET \<br />
-m hashlimit &#8211;hashlimit $RRATE &#8211;hashlimit-burst $BURST \<br />
&#8211;hashlimit-mode srcip,dstport &#8211;hashlimit-name sip_r_limit -j ACCEPT</p>
<p># All other SIP packets&#8230;<br />
$IPTABLES -A sipdos -m hashlimit &#8211;hashlimit $ORATE &#8211;hashlimit-burst $BURST \<br />
&#8211;hashlimit-mode srcip,dstport &#8211;hashlimit-name sip_o_limit -j ACCEPT</p>
<p># Take action on everything else<br />
if [ $LOG ]<br />
then<br />
$IPTABLES -A sipdos -j LOG<br />
fi</p>
<p>$IPTABLES -A sipdos -j $DACTION</p></blockquote>

<div style="font-size:0px;height:0px;line-height:0px;margin:0;padding:0;clear:both"></div>]]></content:encoded>
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		<slash:comments>3</slash:comments>
		</item>
		<item>
		<title>OpenSIPS Externel Perl Script Run</title>
		<link>http://www.howtonix.com/opensips-externel-perl-script-run/</link>
		<comments>http://www.howtonix.com/opensips-externel-perl-script-run/#comments</comments>
		<pubDate>Mon, 07 Jun 2010 17:15:43 +0000</pubDate>
		<dc:creator>emran</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[FreeBSD]]></category>
		<category><![CDATA[Kamailio]]></category>
		<category><![CDATA[OpenSIPS]]></category>
		<category><![CDATA[Perl]]></category>
		<category><![CDATA[Perl Opensips SIPS Redirect Kamailio]]></category>

		<guid isPermaLink="false">http://www.howtonix.com/?p=133</guid>
		<description><![CDATA[You can easily redirect any sip uri based on opensips exec.so module on perl script. you need to load loadmodule &#8220;exec.so&#8221; In the route section you have to exec perl script with paramter $tu (To username) and $fu (from Username). route { if(method == &#8220;INVITE&#8221;) { exec_dset(&#8220;/usr/local/bin/route.pl $fU $tU&#8221;); sl_send_reply(&#8220;302&#8243;,&#8221;LCR Redirect&#8221;); } else { route(1); [...]]]></description>
			<content:encoded><![CDATA[<p>You can easily redirect any sip uri based on opensips exec.so module on perl script. you need to load</p>
<blockquote><p>loadmodule &#8220;exec.so&#8221;</p></blockquote>
<p>In the route section you have to exec perl script with paramter $tu (To username) and $fu (from Username).</p>
<blockquote><p>route {</p>
<p>if(method == &#8220;INVITE&#8221;) {<br />
exec_dset(&#8220;/usr/local/bin/route.pl $fU $tU&#8221;);<br />
sl_send_reply(&#8220;302&#8243;,&#8221;LCR Redirect&#8221;);<br />
} else {<br />
route(1);<br />
}</p>
<p>}</p></blockquote>
<p>In the perl script just write the follow code:</p>
<blockquote><p>#!/bin/sh<br />
echo sip:+121242415112@howtonix.com</p></blockquote>
<p><strong>NOTE: To get this to work correctly with Asterisk you need to add &#8220;promiscredir=yes&#8221; to the general section of your sip.conf<br />
</strong></p>
]]></content:encoded>
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		<slash:comments>1</slash:comments>
		</item>
		<item>
		<title>AMR Codec for Asterisk 1.4 and 1.6</title>
		<link>http://www.howtonix.com/amr-codec-for-asterisk-1-4-and-1-6/</link>
		<comments>http://www.howtonix.com/amr-codec-for-asterisk-1-4-and-1-6/#comments</comments>
		<pubDate>Tue, 29 Sep 2009 19:27:38 +0000</pubDate>
		<dc:creator>emran</dc:creator>
				<category><![CDATA[Asterisk]]></category>

		<guid isPermaLink="false">http://www.howtonix.com/?p=64</guid>
		<description><![CDATA[This patch adds AMR-NB support to Asterisk 1.4 (for Asterisk 1.6 check out asterisk 1.6 branch and use the asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov)) 1. create the asterisk directory $ mkdir asterisk-1.4-SVN-AMR $ cd asterisk-1.4-SVN-AMR/ 2. checkout fontventa repository $ svn checkout http://sip.fontventa.com/svn/asterisk/ fontventa 3. checkout Asterisk $ svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk $ cd [...]]]></description>
			<content:encoded><![CDATA[<p>This patch adds AMR-NB support to Asterisk 1.4</p>
<p>(for Asterisk 1.6 check out asterisk 1.6 branch and use the<br />
asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov))</p>
<p>1. create the asterisk directory</p>
<p>$ mkdir asterisk-1.4-SVN-AMR<br />
$ cd asterisk-1.4-SVN-AMR/</p>
<p>2. checkout fontventa repository</p>
<p>$ svn checkout http://sip.fontventa.com/svn/asterisk/ fontventa</p>
<p>3. checkout Asterisk</p>
<p>$ svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk<br />
$ cd asterisk/</p>
<p>4. add AMR to Asterisk</p>
<p>$ patch &#8211;dry-run -p0 &lt; ../fontventa/amr/amr-asterisk-patch.txt<br />
$ patch -p0 &lt; ../fontventa/amr/amr-asterisk-patch.txt<br />
$ cd codecs<br />
$ ln -s ../../fontventa/amr/amr_slin_ex.h<br />
$ ln -s ../../fontventa/amr/slin_amr_ex.h<br />
$ ln -s ../../fontventa/amr/codec_amr.c<br />
$ mkdir amr<br />
$ cd amr<br />
$ wget http://www.3gpp.org/ftp/Specs/archive/26_series/26.104/26104-700.zip<br />
$ unzip -j 26104-700.zip<br />
$ unzip -j 26104-700_ANSI_C_source_code.zip<br />
$ ln -s ../../../fontventa/amr/Makefile<br />
$ cd ../..</p>
<p>5. build Asterisk</p>
<p>$ ./configure<br />
$ make</p>
<p>6. configure AMR<br />
app_h324m and app_mp4 encode AMR inside the ast_frame in RTP octed aligned mode. (RFC 4867 section 4.4).<br />
To configure the AMR codec to use octed aligned mode add this to codecs.conf:</p>
<p>[amr]<br />
octet-aligned=1</p>
]]></content:encoded>
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		<slash:comments>267</slash:comments>
		</item>
		<item>
		<title>How to Install ztdummy timer for CentOS</title>
		<link>http://www.howtonix.com/how-to-install-ztdummy-timer-for-centos/</link>
		<comments>http://www.howtonix.com/how-to-install-ztdummy-timer-for-centos/#comments</comments>
		<pubDate>Tue, 29 Sep 2009 02:40:12 +0000</pubDate>
		<dc:creator>emran</dc:creator>
				<category><![CDATA[Asterisk]]></category>

		<guid isPermaLink="false">http://www.howtonix.com/?p=58</guid>
		<description><![CDATA[Install the CentOS kernel sources (google it) Build the kernel source Make sure /usr/src/kernel/.config is your kernel config Make sure /lib/modules/`uname -r`/build points to your kernel directory Required Packages: Subversion ncurses ncurses-devel newt newt-devel bison bison-devel cd /usr/src svn co http://svn.digium.com/svn/zaptel/tags/1.4.2.1 zaptel cd zaptel ./configure make make install make config vi /etc/init.d/zaptel change     this: action &#8221;Removing zaptel module: &#8221; rmmod zaptel     to: action &#8221;Removing zaptel module: &#8221; rmmod ztdummy zaptel [...]]]></description>
			<content:encoded><![CDATA[<p>Install the CentOS kernel sources (google it)<br />
Build the kernel source<br />
Make sure /usr/src/kernel/.config is your kernel config<br />
Make sure /lib/modules/`uname -r`/build points to your kernel directory</p>
<p>Required Packages:</p>
<ol>
<li>Subversion</li>
<li>ncurses</li>
<li>ncurses-devel</li>
<li>newt</li>
<li>newt-devel</li>
<li>bison</li>
<li>bison-devel</li>
</ol>
<p>cd /usr/src<br />
svn co http://svn.digium.com/svn/zaptel/tags/1.4.2.1 zaptel<br />
cd zaptel<br />
./configure<br />
make<br />
make install<br />
make config<br />
vi /etc/init.d/zaptel<br />
change<br />
<span style="font-family: monospace;">    this: action &#8221;Removing zaptel module: &#8221; rmmod zaptel </span><br />
<span style="font-family: monospace;">    to: action &#8221;Removing zaptel module: &#8221; rmmod ztdummy zaptel </span><br />
mkdir -p /var/lib/digium/licenses/<br />
useradd asterisk<br />
echo zap/*:asterisk:asterisk:660 &gt;&gt; /etc/udev/permissions.d/50-udev.permissions<br />
/sbin/udevstart<br />
/etc/init.d/zaptel start<br />
chkconfig &#8211;add zaptel</p>
<h3>/etc/init.d/zaptel</h3>
<p>If you are running a RedHat based system run<br />
<span style="font-family: monospace;">  $ make config </span><br />
from within the zaptel source directory. This will create a file called zaptel in /etc/init.d/. Edit the file and search or MODULES and RMODULES, you will see a list of all of the zaptel modules i.e. wcfxo and wctdm(OLD:wcfxs). I deleted all of the entires with the quotes &#8220;&#8221; becasue I didn&#8217;t have any additional hardware and added ztdummy to both fields. You can probably just add ztdummy to the list.</p>
<p>Now if you type<br />
<span style="font-family: monospace;">   $ sudo service zaptel start </span><br />
ztdummy should be running. If you use the ntsysv or similar gui utility you can ensure the service starts after all reboots.</p>
]]></content:encoded>
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		<slash:comments>5</slash:comments>
		</item>
		<item>
		<title>how to send SIP traffic Flood by sipsak?</title>
		<link>http://www.howtonix.com/how-to-send-sip-traffic-flood-by-sipsak/</link>
		<comments>http://www.howtonix.com/how-to-send-sip-traffic-flood-by-sipsak/#comments</comments>
		<pubDate>Sat, 26 Sep 2009 04:50:12 +0000</pubDate>
		<dc:creator>emran</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[FreeSWITCH]]></category>
		<category><![CDATA[OpenSIPS]]></category>
		<category><![CDATA[Yate]]></category>

		<guid isPermaLink="false">http://www.howtonix.com/?p=26</guid>
		<description><![CDATA[sipsak -F -vvv -u 500 -H 192.168.11.33 -l 5060 -s sip:192.168.11.52]]></description>
			<content:encoded><![CDATA[<p><strong>sipsak -F -vvv -u 500 -H 192.168.11.33 -l 5060 -s sip:192.168.11.52</strong></p>
]]></content:encoded>
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		<slash:comments>3</slash:comments>
		</item>
		<item>
		<title>How to capture SIP Packet with Tcpdump ?</title>
		<link>http://www.howtonix.com/how-to-capture-sip-packet-with-tcpdump/</link>
		<comments>http://www.howtonix.com/how-to-capture-sip-packet-with-tcpdump/#comments</comments>
		<pubDate>Sat, 26 Sep 2009 04:49:18 +0000</pubDate>
		<dc:creator>emran</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[FreeSWITCH]]></category>
		<category><![CDATA[OpenSIPS]]></category>
		<category><![CDATA[Yate]]></category>

		<guid isPermaLink="false">http://www.howtonix.com/?p=24</guid>
		<description><![CDATA[tcpdump -i eth0 -n -s0 -v udp port 5060]]></description>
			<content:encoded><![CDATA[<p>tcpdump -i eth0 -n -s0 -v udp port 5060</p>
]]></content:encoded>
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		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Asterisk SS7 Installation</title>
		<link>http://www.howtonix.com/asterisk-ss7-installation/</link>
		<comments>http://www.howtonix.com/asterisk-ss7-installation/#comments</comments>
		<pubDate>Sat, 26 Sep 2009 04:46:26 +0000</pubDate>
		<dc:creator>emran</dc:creator>
				<category><![CDATA[Asterisk]]></category>

		<guid isPermaLink="false">http://www.howtonix.com/?p=17</guid>
		<description><![CDATA[1. install &#38; download Dahdi Complete package: wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz make make install 2. install &#38; download libss7 wget http://downloads.asterisk.org/pub/telephony/libss7/libss7-1.0.2.tar.gz make make install 3. download &#38; install asterisk-1.6.0.14 wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.0.14.tar.gz ./configure make make install now edit /etc/dahdi/system.conf span=1,1,0,ccs,hdb3 bchan=2-31 mtp2=1 echocanceller=mg2,2-31 span=2,2,0,ccs,hdb3 bchan=32-62 echocanceller=mg2,32-62 span=3,3,0,ccs,hdb3 bchan=63-93 echocanceller=mg2,63-93 span=4,4,0,ccs,hdb3 bchan=94-124 echocanceller=mg2,94-124 loadzone = us defaultzone = us [...]]]></description>
			<content:encoded><![CDATA[<p>1. install &amp; download Dahdi Complete package:</p>
<p>wget <a href="http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz">http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz</a></p>
<p>make<br />
make install</p>
<p>2. install &amp; download libss7</p>
<p>wget <a href="http://downloads.asterisk.org/pub/telephony/libss7/libss7-1.0.2.tar.gz">http://downloads.asterisk.org/pub/telephony/libss7/libss7-1.0.2.tar.gz</a></p>
<p>make<br />
make install<br />
3. download &amp; install asterisk-1.6.0.14</p>
<p>wget <a href="http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.0.14.tar.gz">http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.0.14.tar.gz</a></p>
<p>./configure<br />
make<br />
make install</p>
<p>now edit /etc/dahdi/system.conf</p>
<div><span style="font-size: x-small;">span=1,1,0,ccs,hdb3</span></div>
<p><span style="font-size: x-small;">bchan=2-31</p>
<p>mtp2=1</p>
<p>echocanceller=mg2,2-31</p>
<p>span=2,2,0,ccs,hdb3</p>
<p>bchan=32-62</p>
<p>echocanceller=mg2,32-62</p>
<p>span=3,3,0,ccs,hdb3</p>
<p>bchan=63-93</p>
<p>echocanceller=mg2,63-93</p>
<p>span=4,4,0,ccs,hdb3</p>
<p>bchan=94-124</p>
<p>echocanceller=mg2,94-124</p>
<p>loadzone = us</p>
<p>defaultzone = us</p>
<p> </p>
<p></span></p>
<p>finally edit chan_dahdi.conf from /etc/asterisk:</p>
<div><span style="font-size: x-small;">[channels]</span></div>
<p><span style="font-size: x-small;">language=en</p>
<p>context=ss7</p>
<p>switchtype=euroisdn</p>
<p>signalling=ss7</p>
<p>toneduration=100</p>
<p>usecallerid=yes</p>
<p>callwaiting=yes</p>
<p>usecallingpres=yes</p>
<p>callwaitingcallerid=yes</p>
<p>threewaycalling=yes</p>
<p>transfer=yes</p>
<p>canpark=yes</p>
<p>cancallforward=yes</p>
<p>callreturn=yes</p>
<p>echocancel=yes</p>
<p>echocancelwhenbridged=yes</p>
<p>group= 1</p>
<p>callgroup=1</p>
<p>pickupgroup=1</p>
<p>ss7type = itu</p>
<p>ss7_called_nai=dynamic</p>
<p>ss7_calling_nai=dynamic</p>
<p>linkset = 1</p>
<p>pointcode = 1024</p>
<p>adjpointcode = 1234</p>
<p>defaultdpc = 1234</p>
<p>networkindicator=national</p>
<p>cicbeginswith = 2</p>
<p>mtp2=1</p>
<p>sigchan = 1</p>
<p>channel = 2-31</p>
<p>cicbeginswith = 32</p>
<p>channel =33-62</p>
<p>cicbeginswith = 63</p>
<p>channel =63-93</p>
<p>cicbeginswith = 94</p>
<p>channel = 94-124</p>
<p> </p>
<p></span></p>
]]></content:encoded>
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		<slash:comments>4</slash:comments>
		</item>
		<item>
		<title>Free SIP Out Bound Proxy Service</title>
		<link>http://www.howtonix.com/free-sip-out-bound-proxy-service/</link>
		<comments>http://www.howtonix.com/free-sip-out-bound-proxy-service/#comments</comments>
		<pubDate>Sat, 26 Sep 2009 04:36:34 +0000</pubDate>
		<dc:creator>emran</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[FreeSWITCH]]></category>
		<category><![CDATA[OpenSIPS]]></category>
		<category><![CDATA[Yate]]></category>

		<guid isPermaLink="false">http://www.howtonix.com/?p=14</guid>
		<description><![CDATA[Your free SIP Outbound Proxy Service, bypass local  ISPs&#8217; blocking, start using your broadband phone. http://freesps.googlepages.com/]]></description>
			<content:encoded><![CDATA[<p>Your free SIP Outbound Proxy Service, <strong><span style="color: #ff0000;">bypass local  ISPs&#8217; blocking</span></strong>, start using your broadband phone.</p>
<p><a href="http://freesps.googlepages.com/">http://freesps.googlepages.com/</a></p>
]]></content:encoded>
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		<slash:comments>4</slash:comments>
		</item>
	</channel>
</rss>

