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        <title>VOIP-info.org Wiki Changes</title>
        <description><![CDATA[RSS feed for changes to www.voip-info.org wiki pages]]></description>
        <link>http://www.voip-info.org/wiki/</link>
        <lastBuildDate>Sat, 06 Aug 2011 02:40:09 +0100</lastBuildDate>
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        <item>
            <title>remote computer repair</title>
            <link>http://www.voip-info.org/wiki/view/remote+computer+repair</link>
            <description>&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.drpcrepair.com' );&quot;        href='http://www.drpcrepair.com'&gt;online computer repair&lt;/a&gt;&lt;br /&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.drpcrepair.com/remotecomputerrepair.php' );&quot;        href='http://www.drpcrepair.com/remotecomputerrepair.php'&gt;online pc repair&lt;/a&gt;&lt;br /&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.doctorpcrepair.com' );&quot;        href='http://www.doctorpcrepair.com'&gt;remote computer repair&lt;/a&gt;&lt;br /&gt;</description>
            <author>pillatink</author>
            <pubDate>Sat, 06 Aug 2011 08:17:09 +0100</pubDate>
        </item>
        <item>
            <title>Asterisk</title>
            <link>http://www.voip-info.org/wiki/view/Asterisk</link>
            <description>&lt;div class=&quot;maketoc&quot; &gt;&lt;h3&gt;Page Contents&lt;/h3&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#DailyAsteriskNews&quot;&gt;Daily Asterisk News&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#VoIPTodayNews&quot;&gt;VoIP Today News&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#OtherAsteriskNews&quot;&gt;Other Asterisk News&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#StartingOut&quot;&gt;Starting Out&lt;/a&gt;&lt;ul&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#Books&quot;&gt;Books&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Introduction&quot;&gt;Introduction&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Hardware&quot;&gt;Hardware&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Administrationandsystemlayout&quot;&gt;Administration and system layout&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Configuration&quot;&gt;Configuration&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Management&quot;&gt;Management&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Troubleshooting&quot;&gt;Troubleshooting&lt;/a&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#GeneralReference&quot;&gt;General Reference&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;li&gt;&lt;a href=&quot;#CountrySpecificInformation&quot;&gt;Country-Specific Information&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Commercialsupport&quot;&gt;Commercial support&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#SIPServiceProviders&quot;&gt;SIP Service Providers&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#UserGroups&quot;&gt;User Groups&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#WeeklySIPAsteriskUsersConference&quot;&gt;Weekly SIP Asterisk Users Conference&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#InterestingandUnusualProjects&quot;&gt; Interesting and Unusual Projects&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#HowtosandTutorials&quot;&gt;Howtos and Tutorials&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/div&gt;&lt;br /&gt;&lt;img alt=&quot;Image&quot; title=&quot;Image&quot; src=&quot;/img/wiki_up//asterisk-logo.gif&quot; style=&quot;&quot; /&gt;&lt;br /&gt;Original Website - &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.asterisk.org/' );&quot;        href='http://www.asterisk.org/'&gt;http://www.asterisk.org/&lt;/a&gt;&lt;br /&gt;&lt;br /&gt;&lt;strong&gt;Asterisk is a complete PBX in software.&lt;/strong&gt; It runs on &lt;a title=&quot;Linux&quot; href=&quot;/wiki/view/Linux&quot;&gt;Linux&lt;/a&gt;, &lt;a title=&quot;BSD&quot; href=&quot;/wiki/view/BSD&quot;&gt;BSD&lt;/a&gt;, &lt;a title=&quot;Windows&quot; href=&quot;/wiki/view/Windows&quot;&gt;Windows&lt;/a&gt; (emulated) and &lt;a title=&quot;MacOS X&quot; href=&quot;/wiki/view/MacOS+X&quot;&gt;OS X&lt;/a&gt; and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.&lt;br /&gt;&lt;br /&gt;Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, &lt;a title=&quot;ADSI&quot; href=&quot;/wiki/view/ADSI&quot;&gt;ADSI&lt;/a&gt;, &lt;a title=&quot;IAX&quot; href=&quot;/wiki/view/IAX&quot;&gt;IAX&lt;/a&gt;, &lt;a title=&quot;SIP&quot; href=&quot;/wiki/view/SIP&quot;&gt;SIP&lt;/a&gt;, &lt;a title=&quot;H.323&quot; href=&quot;/wiki/view/H.323&quot;&gt;H.323&lt;/a&gt; (as both client and gateway), &lt;a title=&quot;MGCP&quot; href=&quot;/wiki/view/MGCP&quot;&gt;MGCP&lt;/a&gt; (call manager only) and &lt;a title=&quot;SCCP&quot; href=&quot;/wiki/view/SCCP&quot;&gt;SCCP&lt;/a&gt;/Skinny. Check the Features section for a more complete list.&lt;br /&gt;&lt;br /&gt;Asterisk  &lt;strong&gt;needs no additional hardware for Voice-over-IP&lt;/strong&gt;, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism (for certain applications such as conferencing). A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers)&lt;br /&gt;&lt;br /&gt;For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsor, &lt;a title=&quot;Digium&quot; href=&quot;/wiki/view/Digium&quot;&gt;Digium&lt;/a&gt;. ...</description>
            <author>admin</author>
            <pubDate>Sat, 06 Aug 2011 07:56:49 +0100</pubDate>
        </item>
        <item>
            <title>IAXtel.com</title>
            <link>http://www.voip-info.org/wiki/view/IAXtel.com</link>
            <description>IAXtel.com&lt;br /&gt;&lt;br /&gt;Quoted from: &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/iaxtel.com/' );&quot;        href='http://iaxtel.com/'&gt;http://iaxtel.com/&lt;/a&gt;&lt;br /&gt;&lt;br /&gt;Why are there two pages on here for IAXTel?&lt;br /&gt;&lt;br /&gt;Note: This service has been unstable lately. As a noob, do not use it for testing. You may want to try connecting to &lt;a title=&quot;Free World Dialup&quot; href=&quot;/wiki/view/Free+World+Dialup&quot;&gt;Free World Dialup&lt;/a&gt; via IAX2 instead, as it's much more reliable.&lt;br /&gt;&lt;br /&gt;&lt;div class=&quot;bitbox&quot;&gt;&lt;br /&gt;Iaxtel.com allows Asterisk users and IAX clients to connect with each other over the Inter-Asterisk eXchange protocol. Once registered with IAXtel, each user get a unique 1.700 telephone number that will ring their IAX compatible client from any where on the Asterisk network.&lt;br /&gt;&lt;br /&gt;IAXtel is a service offered by &lt;a title=&quot;Digium&quot; href=&quot;/wiki/view/Digium&quot;&gt;Digium&lt;/a&gt;, the sponsors and primary developers of the Asterisk Private Branch Exchange server and the IAX protocol, as well as the GTK based Gnophone GTK based IAX telephone. The IAXtel network is primarily used by Asterisk developers and hobbyists to test and use their systems and study VoIP. Services exist to clear 1.800 calls as well.&lt;br /&gt;&lt;/div&gt;&lt;br /&gt;&lt;br /&gt;</description>
            <author>thegiant</author>
            <pubDate>Fri, 05 Aug 2011 19:43:58 +0100</pubDate>
        </item>
        <item>
            <title>AT-620</title>
            <link>http://www.voip-info.org/wiki/view/AT-620</link>
            <description>&lt;h1&gt;IP Phone ：AT-620&lt;/h1&gt;&lt;span style=&quot;color:red;&quot;&gt;Standard Business IP phone with Broadcom chipset&lt;/span&gt;&lt;br /&gt;&lt;br /&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.atcom.cn/high%20resolution/AT620-1.jpg' );&quot;        href='http://www.atcom.cn/high%20resolution/AT620-1.jpg'&gt;&lt;img alt=&quot;Image&quot; title=&quot;Image&quot; src=&quot;http://www.atcom.cn/products_gif/AT-620.jpg&quot; style=&quot;&quot; /&gt; &lt;/a&gt;&lt;br /&gt;&lt;br /&gt;&lt;span style=&quot;font-family:monospace;&quot;&gt;AT620&amp;nbsp;VoIP&amp;nbsp;phone&amp;nbsp;is&amp;nbsp;ATCOM&amp;nbsp;business&amp;nbsp;IP&amp;nbsp;phone&amp;nbsp;terminal&amp;nbsp;with&amp;nbsp;Broadcom&amp;nbsp;solution&amp;nbsp;which&amp;nbsp;adopts&amp;nbsp;multiple&amp;nbsp;voice&amp;nbsp;control&amp;nbsp;protocols&amp;nbsp;and&amp;nbsp;voice&amp;nbsp;compression&amp;nbsp;codec&amp;nbsp;to&amp;nbsp;directly&amp;nbsp;convert&amp;nbsp;analog&amp;nbsp;voice&amp;nbsp;into&amp;nbsp;IP&amp;nbsp;packet&amp;nbsp;for&amp;nbsp;internet&amp;nbsp;transport,&amp;nbsp;thus&amp;nbsp;effectively&amp;nbsp;using&amp;nbsp;the&amp;nbsp;existing&amp;nbsp;bandwidth&amp;nbsp;to&amp;nbsp;provide&amp;nbsp;PSTN&amp;nbsp;quality&amp;nbsp;voice&amp;nbsp;service.&lt;/span&gt;&lt;br /&gt;AT620 IP phone supports SIP and IAX2 protocol, offering two 10/100Mbps Ethernet interface with built in router. It is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.&lt;br /&gt;&lt;br /&gt;&lt;h2&gt;Features:&lt;/h2&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;VoIP&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;SIP (SIP RFC3261,RFC 2543)/IAX2 support.&lt;/li&gt;&lt;li&gt;2 SIP lines + 1 IAX2 line&lt;/li&gt;&lt;li&gt;Redundancy sip server capable.&lt;/li&gt;&lt;li&gt;Hotline.&lt;/li&gt;&lt;li&gt;Call Forward、Call transfer、Call hold、Call waiting, &lt;/li&gt;&lt;li&gt;3-way Talking、Pickup、Join call、Redial、Unredial、&lt;/li&gt;&lt;li&gt;Call Park、vport、click to dial&lt;/li&gt;&lt;li&gt;DND(Do Not Disturb),&lt;/li&gt;&lt;li&gt;Black List,Limit List&lt;/li&gt;&lt;li&gt;E.164 dial plan and customized dial rules &lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Voice&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;Tone generation and Local DTMF re-generation according with ITU-T&lt;/li&gt;&lt;li&gt;G.711(A-law or u-law) , G722 , G723 G729&lt;/li&gt;&lt;li&gt;AGC(Auto Gain Control)&lt;/li&gt;&lt;li&gt;AEC(Auto Echo Cancellation)&lt;/li&gt;&lt;li&gt;VAD (Voice Activity Detection)&lt;/li&gt;&lt;li&gt;CNG(Comfort Noise Generation&lt;/li&gt;&lt;li&gt;G.165 compliant 96 ms echo cancellation&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Networking&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;Static/Dynamic WAN-IP-Addressing&lt;/li&gt;&lt;li&gt;NAT, Firewall.&lt;/li&gt;&lt;li&gt;Support VPN (L2TP) Open VPN&lt;/li&gt;&lt;li&gt;Support DMZ&lt;/li&gt;&lt;li&gt;WAN support Primary and Alter function&lt;/li&gt;&lt;li&gt;Qos support Diffserv&lt;/li&gt;&lt;li&gt;Support VLAN&lt;/li&gt;&lt;li&gt;DHCP client and server.&lt;/li&gt;&lt;li&gt;Support PPPoE, (used for ADSL, cable modem connecting).&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Protocols&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;MAC Address&lt;/li&gt;&lt;li&gt;TCP:Transmission Control Protocol &lt;/li&gt;&lt;li&gt;DHCP:Dynamic Host Configuration Protocol&lt;/li&gt;&lt;li&gt;PPPoE:PPP Protocol over Ethernet&lt;/li&gt;&lt;li&gt;PoE(option)&lt;/li&gt;&lt;li&gt;SNTP, Simple Network Time Protocol&lt;/li&gt;&lt;li&gt;STUN - Simple Traversal of User Datagram ...&lt;/li&gt;&lt;li&gt;MD5 Message-Digest Algorithm&lt;/li&gt;&lt;li&gt;DNS: Domain Name Server&lt;/li&gt;&lt;li&gt;RTP: Real-time Transport Protocol&lt;/li&gt;&lt;li&gt;RTCP:Real-time Control Protocol&lt;/li&gt;&lt;li&gt;Telnet:Internet's remote login protocol&lt;/li&gt;&lt;li&gt;HTTP:Hyper Text Transfer protocol&lt;/li&gt;&lt;li&gt;FTP:File Transfer protocol&lt;/li&gt;&lt;li&gt;TFTP:Trivial File Transfer Protocol&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Management&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;Web, telnet and keypad management.&lt;/li&gt;&lt;li&gt;Adjustable user password and super password&lt;/li&gt;&lt;li&gt;Upgrade firmware through HTTP, FTP or TFTP.&lt;/li&gt;&lt;li&gt;Telnet remote management. ...</description>
            <author>michaelj</author>
            <pubDate>Fri, 05 Aug 2011 18:55:34 +0100</pubDate>
        </item>
        <item>
            <title>AT-530</title>
            <link>http://www.voip-info.org/wiki/view/AT-530</link>
            <description>&lt;h1&gt;&lt;span style=&quot;color:blue;&quot;&gt;IP Phone ：AT-530&lt;/span&gt;&lt;/h1&gt;&lt;ul&gt;&lt;li&gt;&lt;span style=&quot;color:red;&quot;&gt;High performance IP phone with Infineon solution&lt;/span&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.atcom.cn/high%20resolution/AT530-1.jpg' );&quot;        href='http://www.atcom.cn/high%20resolution/AT530-1.jpg'&gt;&lt;img alt=&quot;Image&quot; title=&quot;Image&quot; src=&quot;http://www.atcom.cn/products_gif/AT-530.jpg&quot; style=&quot;&quot; /&gt;&lt;/a&gt;&lt;br /&gt;&lt;br /&gt;AT530 VoIP phone is a business IP phone terminal with high performance Infineon solution which adopts multiple voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.&lt;br /&gt;AT530 IP phone supports SIP and IAX2 protocol, offering two 10/100Mbps Ethernet interface with built in router. It is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.&lt;br /&gt;&lt;br /&gt;&lt;h2&gt;Features:&lt;/h2&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Key features:&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;SIP/IAX2 support.&lt;/li&gt;&lt;li&gt;Support two sip servers running at the same time.&lt;/li&gt;&lt;li&gt;Redundancy sip server capable.&lt;/li&gt;&lt;li&gt;NAT, Firewall.&lt;/li&gt;&lt;li&gt;DHCP client and server.&lt;/li&gt;&lt;li&gt;Support PPPoE, (used for ADSL, cable modem connecting).&lt;/li&gt;&lt;li&gt;Support major G7.xxx CODEC.&lt;/li&gt;&lt;li&gt;VAD,CNG.&lt;/li&gt;&lt;li&gt;G.165 compliant 16ms echo cancellation&lt;/li&gt;&lt;li&gt;E.164 dial plan and customized dial rules&lt;/li&gt;&lt;li&gt;Hotline.&lt;/li&gt;&lt;li&gt;Call Forward, Call Transfer, 3-way conference calls&lt;/li&gt;&lt;li&gt;Call ID display&lt;/li&gt;&lt;li&gt;DND(Do Not Disturb),Black List,Limit List &lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Data Features:&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;Static/Dynamic WAN-IP-Addressing &lt;/li&gt;&lt;li&gt;PPPoE&lt;/li&gt;&lt;li&gt;POE&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Management：&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;Web, telnet and keypad management.&lt;/li&gt;&lt;li&gt;Adjustable user password and super password&lt;/li&gt;&lt;li&gt;Upgrade firmware through HTTP, FTP or TFTP.&lt;/li&gt;&lt;li&gt;Telnet remote management.&lt;/li&gt;&lt;li&gt;Upload/download setting file&lt;/li&gt;&lt;li&gt;Auto-provision.&lt;/li&gt;&lt;li&gt;Safe mode provide reliability&lt;/li&gt;&lt;li&gt;Phone book, maximum 100 entries.&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Interface：&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;Two RJ45 ports, one for WAN, one LAN.&lt;/li&gt;&lt;li&gt;Power port&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;h2&gt;Specifications:&lt;/h2&gt;&lt;strong&gt;Supported specification and applications&lt;/strong&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Data networking:&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;MAC Address&lt;/li&gt;&lt;li&gt;TCP:Transmission Control Protocol &lt;/li&gt;&lt;li&gt;DHCP:Dynamic Host Configuration Protocol&lt;/li&gt;&lt;li&gt;PPPoE:PPP Protocol over Ethernet&lt;/li&gt;&lt;li&gt;POE(option)&lt;/li&gt;&lt;li&gt;SNTP, Simple Network Time Protocol&lt;/li&gt;&lt;li&gt;STUN - Simple Traversal of User Datagram ...&lt;/li&gt;&lt;li&gt;MD5 Message-Digest Algorithm&lt;/li&gt;&lt;li&gt;DNS：Domain Name Server&lt;/li&gt;&lt;li&gt;RTP: Real-time Transport Protocol&lt;/li&gt;&lt;li&gt;RTCP:Real-time Control Protocol&lt;/li&gt;&lt;li&gt;Telnet:Internet's remote login protocol&lt;/li&gt;&lt;li&gt;HTTP:Hyper Text Transfer protocol&lt;/li&gt;&lt;li&gt;FTP:File Transfer protocol&lt;/li&gt;&lt;li&gt;TFTP:Trivial File Transfer Protocol&lt;/li&gt;&lt;li&gt;POE:Power Over Ethernet &lt;/li&gt;&lt;li&gt;Call control /voip Features&lt;/li&gt;&lt;li&gt;SIP RFC3261,RFC 2543&lt;/li&gt;&lt;li&gt;Tone generation and Local DTMF re-generation according with ITU-T&lt;/li&gt;&lt;li&gt;G.711(A-law or u-law)&lt;/li&gt;&lt;li&gt;G.723.1(6.3kbps,5.3 kbps)&lt;/li&gt;&lt;li&gt;G729&lt;/li&gt;&lt;li&gt;AGC(Auto Gain Control)&lt;/li&gt;&lt;li&gt;G. ...</description>
            <author>michaelj</author>
            <pubDate>Fri, 05 Aug 2011 18:53:04 +0100</pubDate>
        </item>
        <item>
            <title>AX-400P</title>
            <link>http://www.voip-info.org/wiki/view/AX-400P</link>
            <description>&lt;h1&gt;&lt;span style=&quot;color:blue;&quot;&gt;AX-400P&lt;/span&gt;&lt;/h1&gt;&lt;span style=&quot;color:red;&quot;&gt;4 ports FXO/FXS card for Asterisk PBX&lt;/span&gt;&lt;br /&gt;&lt;br /&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.atcom.cn/high%20resolution/AX400P.jpg' );&quot;        href='http://www.atcom.cn/high%20resolution/AX400P.jpg'&gt;&lt;img alt=&quot;Image&quot; title=&quot;Image&quot; src=&quot;http://www.atcom.cn/products_gif/AX-400P.jpg&quot; style=&quot;&quot; /&gt;&lt;/a&gt;&lt;br /&gt;&lt;br /&gt;AX-400P is the TDM400P compatible PCI card that support up to four FXO and FXS combination interfaces. Using AX-400P series PCI card, open source Asterisk PBX and stand alone PC, users can create their SOHO telephony solution include all the sophisticated features of traditional PBX, and extend features such as voicemail in IP PBX.&lt;br /&gt;User can use AX-400P with standard Zaptel device driver and Asterisk source code without modify any code. The FXO and FXS module are interchangeable to suit various requirements.&lt;br /&gt;&lt;br /&gt;&lt;h2&gt;Features&lt;/h2&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Features:&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;Analog card for Asterisk PBX&lt;/li&gt;&lt;li&gt;Support Asterisk PBX and zaptel driver&lt;/li&gt;&lt;li&gt;Support up to four fxo/fxs analog port&lt;/li&gt;&lt;li&gt;Suitable for SOHO PBX / VoiceMail / IVR.&lt;/li&gt;&lt;li&gt;Caller ID and Call waiting Caller ID&lt;/li&gt;&lt;li&gt;Conference&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;h2&gt;Specifications&lt;/h2&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Configuration:&lt;/strong&gt;	&lt;ul&gt;&lt;li&gt;Motherboard: AX-400P&lt;/li&gt;&lt;li&gt;Single port FXS module: AX-110S&lt;/li&gt;&lt;li&gt;Single port FXO module: AX-110X&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;Hardware requirement:&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;500-Mhz Pentium III &lt;/li&gt;&lt;li&gt;64MB RAM&lt;/li&gt;&lt;li&gt;3.3V or 5V PCI 2.2 slot&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;strong&gt;PCI card dimension:&lt;/strong&gt;&lt;ul&gt;&lt;li&gt;102mm (height) × 134mm (Length) &lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;h2&gt;Contact ATCOM:&lt;/h2&gt;ATCOM Technology co., LTD.&lt;br /&gt;Address: A2F , Block 3 ,Huangguan Technology Park , #21 Tairan 9th Rd, Chegongmiao , Futian District , Shenzhen China &lt;br /&gt;Tel: +(86)755-23487618 &lt;br /&gt;Fax: +(86)755-23485319&lt;br /&gt;E-mail: &lt;a  href='&amp;#115;&amp;#097;&amp;#108;&amp;#101;&amp;#115;&amp;#064;&amp;#097;&amp;#116;&amp;#099;&amp;#111;&amp;#109;&amp;#101;&amp;#109;&amp;#097;&amp;#105;&amp;#108;&amp;#046;&amp;#099;&amp;#111;&amp;#109;'&gt;&amp;#115;&amp;#097;&amp;#108;&amp;#101;&amp;#115;&amp;#064;&amp;#097;&amp;#116;&amp;#099;&amp;#111;&amp;#109;&amp;#101;&amp;#109;&amp;#097;&amp;#105;&amp;#108;&amp;#046;&amp;#099;&amp;#111;&amp;#109;&lt;/a&gt;  &lt;br /&gt;Website: &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.atcom.cn' );&quot;        href='http://www.atcom.cn'&gt;http://www.atcom.cn&lt;/a&gt;&lt;br /&gt;&lt;br /&gt;&lt;h2&gt;European Distribution:&lt;/h2&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.ipchitchat.co.uk/' );&quot;        href='http://www.ipchitchat.co.uk/'&gt;IPChit-Chat&lt;/a&gt; UK and European Distributor for &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.ipchitchat.co.uk/index.php/Atcom-Cards' );&quot;        href='http://www.ipchitchat.co.uk/index.php/Atcom-Cards'&gt;Atcom Cards&lt;/a&gt; and &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.ipchitchat.co.uk/index.php/AX400p' );&quot;        href='http://www.ipchitchat.co.uk/index.php/AX400p'&gt;Atcom AX400p&lt;/a&gt;&lt;br /&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.netcloudgroup.co.uk' );&quot;        href='http://www.netcloudgroup.co.uk'&gt;Atcom UK and European Distributor&lt;/a&gt; - Netcloud&lt;br /&gt;&lt;br /&gt;&lt;br /&gt;&lt;br /&gt;&lt;h2&gt;Oceania Distribution:&lt;/h2&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/nicegear.co.nz/' );&quot;        href='http://nicegear.co.nz/'&gt;nicegear - New Zealand's VoIP Hardware Supplier&lt;/a&gt;&lt;br /&gt;&lt;br /&gt;&lt;h2&gt;See Also&lt;/h2&gt;&lt;span style=&quot;font-family:monospace;&quot;&gt;&lt;a title=&quot;Atcom&quot; href=&quot;/wiki/view/Atcom&quot;&gt;ATcom&lt;/a&gt;&lt;/span&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker. ...</description>
            <author>michaelj</author>
            <pubDate>Fri, 05 Aug 2011 18:50:07 +0100</pubDate>
        </item>
        <item>
            <title>Grandstream GXP 2100</title>
            <link>http://www.voip-info.org/wiki/view/Grandstream+GXP+2100</link>
            <description>&lt;h1&gt;&lt;strong&gt;&lt;span style=&quot;color:blue;&quot;&gt;Grandstream GXP2100 HD Enterprise 4-line PoE IP Phone&lt;/span&gt;&lt;/strong&gt;&lt;/h1&gt;&lt;br /&gt;&lt;br /&gt;&lt;div class=&quot;item&quot;&gt;&lt;a href=&quot;/liberty/view/file/3840&quot;&gt;&lt;img class=&quot;thumb&quot; src=&quot;http://www.voip-info.org/storage/users/600/94600/images/3840/medium.jpg&quot; alt=&quot;GRGXP2100.jpg&quot; title=&quot;GRGXP2100.jpg&quot;/&gt;&lt;/a&gt;&lt;/div&gt;&lt;br /&gt;&lt;br /&gt;&lt;br /&gt;The Grandstream GXP2100 IP Phone is a next generation, enterprise grade IP phone that features 4 lines a 180 x 90 backlit graphical LCD, 3 XML programmable context-sensitive soft keys, &lt;br /&gt;7 XML programmable BLF extension keys, dual network ports with integrated PoE and 5 way conferencing.&lt;br /&gt;&lt;br /&gt;The Grandstream GXP2100 IP Phone delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service,&lt;br /&gt;automated provisioning for easy deployment, advanced security protection for privacy and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.&lt;br /&gt;&lt;span style=&quot;font-family:monospace;&quot;&gt;&lt;/span&gt;&lt;br /&gt;It is a perfect choice for enterprise users looking for a high quality, feature rich multi-line IP phone with the best values.&lt;br /&gt;&lt;br /&gt;&lt;h1&gt;&lt;strong&gt;&lt;span style=&quot;color:blue;&quot;&gt;Feature Highlights&lt;/span&gt;&lt;/strong&gt;&lt;/h1&gt;&lt;span style=&quot;font-family:monospace;&quot;&gt;&lt;/span&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;180x90 pixel backlit graphical LCD display with up to 8 level grayscale&lt;/li&gt;&lt;li&gt;Dual switched auto-sensing 10/100Mbps network ports with integrated PoE&lt;/li&gt;&lt;li&gt;Large phonebook (up to 2,000 contacts) and call history (up to 2,000 records)&lt;/li&gt;&lt;li&gt;4 dual-color line keys (with 4 SIP accounts), up to 5-way conference (max 4 concurrent calls), and up to 11 call appearances with 7 dual-color BLF (Busy Lamp Field) extension keys&lt;/li&gt;&lt;li&gt;3 XML programmable context-sensitive soft keys, a desktop stand to allow 2-angle positions&lt;/li&gt;&lt;li&gt;HD wideband audio, superb full-duplex hands-free speakerphone with advanced acoustic echo cancellation and excellent double-talk performance&lt;/li&gt;&lt;li&gt;Automated personal information service (e.g. local weather, stock, currencies, RSS news, etc), personalized music ring tone/music ring back tone/music-on-hold using Internet music streaming or local music files, flexible customizable screen content &amp;amp; format using Web and enterprise applications integration (pending)&lt;/li&gt;&lt;li&gt;RJ9 and 2.5mm jacks for professional and casual use&lt;/li&gt;&lt;li&gt;Can be used as a desktop phone or mounted on a wall&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;&lt;span style=&quot;font-family:monospace;&quot;&gt;&lt;/span&gt;&lt;br /&gt;&lt;h1&gt;&lt;strong&gt;&lt;span style=&quot;color:blue;&quot;&gt;Protocols/Standards&lt;/span&gt;&lt;/strong&gt;&lt;/h1&gt;&lt;span style=&quot;font-family:monospace;&quot;&gt;&lt;/span&gt;&lt;br /&gt;&lt;ul&gt;&lt;li&gt;SIP RFC3261&lt;/li&gt;&lt;li&gt;TCP/IP/UDP&lt;/li&gt;&lt;li&gt;RTP/RTCP&lt;/li&gt;&lt;li&gt;HTTP/HTTPS&lt;/li&gt;&lt;li&gt;ARP/RARP&lt;/li&gt;&lt;li&gt;ICMP&lt;/li&gt;&lt;li&gt;DNS (A record, SRV, NAPTR)&lt;/li&gt;&lt;li&gt;DHCP&lt;/li&gt;&lt;li&gt;PPPoE&lt;/li&gt;&lt;li&gt;TELNET&lt;/li&gt;&lt;li&gt;TFTP&lt;/li&gt;&lt;li&gt;NTP&lt;/li&gt;&lt;li&gt;STUN&lt;/li&gt;&lt;li&gt;SIMPLE&lt;/li&gt;&lt;li&gt;TR-069&lt;/li&gt;&lt;li&gt;802. ...</description>
            <author>michaelj</author>
            <pubDate>Fri, 05 Aug 2011 18:43:01 +0100</pubDate>
        </item>
        <item>
            <title>voip-info.org</title>
            <link>http://www.voip-info.org/wiki/view/voip-info.org</link>
            <description>&lt;h1&gt;Welcome to the VOIP Wiki - a reference guide to all things VOIP&lt;/h1&gt;&lt;br /&gt;This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. &lt;strong&gt;However, the Wiki is primarily for information, not for advertising.&lt;/strong&gt;&lt;br /&gt;&lt;br /&gt;&lt;strong&gt;Your contributions are welcome, &lt;span style=&quot;color:red;&quot;&gt;please read the &lt;a title=&quot;How to add information to this wiki&quot; href=&quot;/wiki/view/How+to+add+information+to+this+wiki&quot;&gt;How to add information to this wiki&lt;/a&gt; page and the &lt;/strong&gt;&lt;a title=&quot;Posting Guidelines for Promoting Products and Services&quot; href=&quot;/wiki/view/Posting+Guidelines+for+Promoting+Products+and+Services&quot;&gt;Posting Guidelines&lt;/a&gt;&lt;/span&gt;&lt;strong&gt; &lt;span style=&quot;color:red;&quot;&gt;before you post.&lt;/span&gt;&lt;/strong&gt;&lt;br /&gt; &lt;br /&gt;&lt;br /&gt;&lt;h2&gt;NEWS&lt;/h2&gt;&lt;ul&gt;&lt;li&gt; 2011-08-03 - &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/sip-router.org/10-years-ser/' );&quot;        href='http://sip-router.org/10-years-ser/'&gt;10 years SER&lt;/a&gt; - open event in Berlin, Sep 2, 2011, to celebrate 10 years of SIP Express Router, the open source SIP server&lt;/li&gt;&lt;li&gt; 2011-08-02 - &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.iia-inc.com/TeleWatch.htm ' );&quot;        href='http://www.iia-inc.com/TeleWatch.htm '&gt; TeleWatch&lt;/a&gt; Plug Computer based Video Surveillance and IP-PBX&lt;/li&gt;&lt;li&gt; 2011-07-26 -  &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.sevana.fi/mailman/listinfo/asteriskvqm_sevana.fi ' );&quot;        href='http://www.sevana.fi/mailman/listinfo/asteriskvqm_sevana.fi '&gt; Asterisk VQM Maillist&lt;/a&gt; Join Asterisk VQM mail list to discuss Asterisk Voice Quality Monitoring solution. Questions, feature requests are welcome!&lt;/li&gt;&lt;li&gt; 2011-07-26 -  &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/nerdvittles.com/' );&quot;        href='http://nerdvittles.com/?p=767 '&gt; Nerd Vittles&lt;/a&gt; introduces Asterisk 10, PBX in a Flash 1.7.5.6.3, Incredible PBX 2.0, and more...&lt;/li&gt;&lt;li&gt; 2011-07-26 -  New model ATA &lt;a  href='http://www.voip-info.org/wiki/view/AG-198'&gt;AG198&lt;/a&gt; with 1 FXS and PSTN passby is launched by ATCOM.&lt;a  href='http://www.voip-info.org/wiki/view/Atcom'&gt;&amp;lt;&amp;lt;more&lt;/a&gt;&lt;/li&gt;&lt;li&gt; 2011-07-23 - &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.ictinnovations.com' );&quot;        href='http://www.ictinnovations.com'&gt;ICT Innovations&lt;/a&gt; released new version of &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/sourceforge.net/projects/ictfax/' );&quot;        href='http://sourceforge.net/projects/ictfax/'&gt;ICTFAX Version 0.3.2&lt;/a&gt; , a complete open source fax over ip and billing solution .&lt;/li&gt;&lt;li&gt; 2011-07-23 - &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/sourceforge.net/projects/openss7gw/files/' );&quot;        href='http://sourceforge.net/projects/openss7gw/files/'&gt;OpenSGW -An Open Source SS7 Signaling Gateway-&lt;/a&gt; project has released a new component: a generic parser for SCTP and M3UA layers config file. The configuration file format is very similar to DIALOGIC SS7 config file.&lt;/li&gt;&lt;li&gt; 2011-07-22 - ATCOM release high end model IP phone &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.atcom.cn/AT640.html' );&quot;        href='http://www.atcom.cn/AT640.html'&gt;AT640&lt;/a&gt; with BLF and attachable extension modules.&lt;a  href='http://www.voip-info.org/wiki/view/Atcom'&gt;&amp;lt;&amp;lt;more&lt;/a&gt;&lt;/li&gt;&lt;li&gt; 2011-07-21 - &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.newfies-dialer.org/' );&quot;        href='http://www.newfies-dialer.org/'&gt;Newfies-Dialer&lt;/a&gt; Version 0.1 released | Open Source Voice Broadcast application for FreeSWITCH&lt;/li&gt;&lt;li&gt; 2011-07-20 - &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.vadaxchange.com' );&quot;        href='http://www.vadaxchange.com'&gt;VadaXchange Buddy&lt;/a&gt; version 1.0 released - new user-friendly call control interface developed on top of an Asterisk PABX. ...</description>
            <author>techsupport</author>
            <pubDate>Fri, 05 Aug 2011 16:44:36 +0100</pubDate>
        </item>
        <item>
            <title>Xorcom Astribank</title>
            <link>http://www.voip-info.org/wiki/view/Xorcom+Astribank</link>
            <description>The Astribank is a USB &lt;a title=&quot;Asterisk Channel Bank&quot; href=&quot;/wiki/view/Asterisk+Channel+Bank&quot;&gt;channel bank&lt;/a&gt; for &lt;a title=&quot;Asterisk&quot; href=&quot;/wiki/view/Asterisk&quot;&gt;Asterisk&lt;/a&gt; developed by &lt;a title=&quot;Xorcom&quot; href=&quot;/wiki/view/Xorcom&quot;&gt;Xorcom&lt;/a&gt;.&lt;br /&gt; &lt;br /&gt;&lt;div class=&quot;maketoc&quot; &gt;&lt;h3&gt;Page Contents&lt;/h3&gt;&lt;ul&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#Overview&quot;&gt; Overview&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Features&quot;&gt; Features&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Installation&quot;&gt;Installation&lt;/a&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#DebianStableRepository&quot;&gt;Debian Stable Repository&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#DebianEtchRepository&quot;&gt;Debian Etch Repository&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#TrixBoxCentOSRepository&quot;&gt;TrixBox/CentOS Repository&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#OtherDistributions&quot;&gt;Other Distributions&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#FromSource&quot;&gt;From Source&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#InitScript&quot;&gt;Init Script&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Configuration&quot;&gt;Configuration&lt;/a&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#ManualTrixboxConfiguration&quot;&gt;Manual Trixbox Configuration&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#MultipleAstribanks&quot;&gt;Multiple Astribanks&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Troubleshooting&quot;&gt;Troubleshooting&lt;/a&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#BuildProblems&quot;&gt;Build Problems&lt;/a&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#OnlyXPPnotBuilt&quot;&gt;Only XPP not Built&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#ErrorBuildingfpga_load&quot;&gt;Error Building fpga_load&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#SoundQuality&quot;&gt;Sound Quality&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Whenallswell&quot;&gt;When all's well:&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#ErrorMessages&quot;&gt;Error Messages&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#AlwaysLastSpan&quot;&gt;Always Last Span&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#InputsandOutputs&quot;&gt;Inputs and Outputs&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#WiringofPorts&quot;&gt;Wiring of Ports&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#TheprocxppInterface&quot;&gt;The /proc/xpp Interface&lt;/a&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#AddingandRemoving&quot;&gt;Adding and Removing&lt;/a&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#AsteriskNotRunning&quot;&gt;Asterisk Not Running&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#WithAsteriskRunning&quot;&gt;With Asterisk Running&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Patches&quot;&gt;Patches&lt;/a&gt;&lt;ul&gt;&lt;li&gt;&lt;a href=&quot;#zap_restart&quot;&gt;zap_restart&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#zt_event_removed&quot;&gt;zt_event_removed&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#autoztcfg&quot;&gt;autoztcfg&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#XPPandXPD&quot;&gt;XPP and XPD&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Seealso&quot;&gt;See also&lt;/a&gt;&lt;/li&gt;&lt;li&gt;&lt;a href=&quot;#Distributors&quot;&gt;Distributors:&lt;/a&gt;&lt;/li&gt;&lt;/ul&gt;&lt;/ul&gt;&lt;/div&gt;&lt;br /&gt;&lt;div class=&quot;item&quot;&gt;&lt;a href=&quot;/liberty/view/file/3444&quot;&gt;&lt;img class=&quot;thumb&quot; src=&quot;http://www.voip-info.org/storage/users/740/17740/images/3444/medium.png&quot; alt=&quot;astribank-angled.png&quot; title=&quot;astribank-angled.png&quot;/&gt;&lt;/a&gt;&lt;/div&gt;&lt;br /&gt;see: &lt;br /&gt;&lt;strong&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.xorcom.com/astribank-technology/astribank-vs-channel-banks.html' );&quot;        href='http://www.xorcom.com/astribank-technology/astribank-vs-channel-banks.html'&gt;Astribank vs. Legacy Channel Banks&lt;/a&gt;&lt;/strong&gt;&lt;br /&gt;&lt;strong&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.xorcom.com/astribank-technology/astribank-vs-pci-cards.html' );&quot;        href='http://www.xorcom.com/astribank-technology/astribank-vs-pci-cards.html'&gt;Astribank vs. PCI Cards&lt;/a&gt;&lt;/strong&gt;&lt;br /&gt;&lt;strong&gt;&lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.xorcom.com/astribank-technology/astribank-vs-ip-gateways.html' );&quot;        href='http://www.xorcom.com/astribank-technology/astribank-vs-ip-gateways.html'&gt;Astribank vs. ...</description>
            <author>nathansinton</author>
            <pubDate>Fri, 05 Aug 2011 16:29:22 +0100</pubDate>
        </item>
        <item>
            <title>chan_mobile</title>
            <link>http://www.voip-info.org/wiki/view/chan_mobile</link>
            <description>&lt;h1&gt;chan_mobile (used to be chan_cellphone) &amp;mdash; Use Bluetooth cell / mobile phones as FXO devices&lt;/h1&gt;&lt;br /&gt;Asterisk Channel Driver to allow Bluetooth Cell/Mobile Phones to be used as FXO devices and Bluetooth Headsets as FXS devices&lt;br /&gt;The official Homepage is &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.chan-mobile.org' );&quot;        href='http://www.chan-mobile.org'&gt;http://www.chan-mobile.org&lt;/a&gt; (no longer available).&lt;br /&gt;&lt;span style=&quot;font-family:monospace;&quot;&gt;&lt;/span&gt;&lt;br /&gt;Features :- (Oct 2007)&lt;br /&gt;&lt;ul&gt;&lt;li&gt; Multiple cell phones can be connected (subject to some limitations - see Notes).&lt;/li&gt;&lt;li&gt; Multiple bluetooth adapters can be supported.&lt;/li&gt;&lt;li&gt; Asterisk automatically connects to each cell phone when it comes in range.&lt;/li&gt;&lt;li&gt; Command to discover bluetooth devices. Useful for configuration. Requires an unused bluetooth adapter.&lt;/li&gt;&lt;li&gt; Inbound calls to the cell phones are handled by Asterisk, just like inbound calls on a Zap channel.&lt;/li&gt;&lt;li&gt; Caller ID passed through on inbound calls.&lt;/li&gt;&lt;li&gt; Dial outbound on a cell phone using Dial(CELL/device/nnnnnnn) in the dialplan.&lt;/li&gt;&lt;li&gt; Use a Bluetooth Headset as extension using Dial(CELL/device) in the dialplan. &lt;/li&gt;&lt;li&gt; Application CellStatus can be used in the dialplan to see if a cell phone is connected.&lt;/li&gt;&lt;li&gt; Application MobileSMS to send SMS via a connected mobile phone&lt;/li&gt;&lt;li&gt; Supports devicestate for dialplan hinting.&lt;/li&gt;&lt;/ul&gt;&lt;br /&gt;chan_mobile was written by David Bowerman and is officially supported only for the Asterisk development trunk and is available as an add-on from &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/svnview.digium.com/svn/asterisk/trunk/addons/chan_mobile.c' );&quot;        href='http://svnview.digium.com/svn/asterisk/trunk/addons/chan_mobile.c?view=markup'&gt;http://svnview.digium.com/svn/asterisk/trunk/addons/chan_mobile.c?view=markup&lt;/a&gt;. Current documentation can be obtained from &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/svnview.digium.com/svn/asterisk-addons/branches/1.6.2/doc/chan_mobile.txt' );&quot;        href='http://svnview.digium.com/svn/asterisk-addons/branches/1.6.2/doc/chan_mobile.txt?view=markup'&gt;http://svnview.digium.com/svn/asterisk-addons/branches/1.6.2/doc/chan_mobile.txt?view=markup&lt;/a&gt;&lt;br /&gt;Unofficial and unsupported Asterisk 1.2 backport can be obtained from &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.sigsegv.cx/sip-9.html' );&quot;        href='http://www.sigsegv.cx/sip-9.html'&gt;http://www.sigsegv.cx/sip-9.html&lt;/a&gt;.&lt;br /&gt;A good installation guide for trixbox can be found under &lt;a class=&quot;external&quot; onClick=&quot;javascript: pageTracker._trackPageview('/outgoing/wikipages/www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html' );&quot;        href='http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html'&gt;http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html&lt;/a&gt;.&lt;br /&gt;&lt;strong&gt;Note:&lt;/strong&gt; FC6 bluez yum packages with latest updates will not work, see comment below&lt;br /&gt;&lt;strong&gt;Note:&lt;/strong&gt; Some mobiles (Motorola V3 and K1) report themselves as a valid headset, but  they do not work if you configure them as a headset in mobile.conf (not fully tested)&lt;br /&gt;&lt;strong&gt;Note:&lt;/strong&gt; Not all mobiles with bluetooth profiles have all the features necessary for this channel to work. For example - Nokia E65 is not usable.&lt;br /&gt;&lt;strong&gt;Note:&lt;/strong&gt; Each mobile &quot;eats&quot; one bluetooth adapter. Multiple mobiles cannot connect to the same adapter so if you want to connect multiple mobiles prepare to buy dongles by the basket. &lt;br /&gt;&lt;br /&gt;&lt;h2&gt;SMS&lt;/h2&gt;In chan_mobile.c, you'll see apps MobileSendSMS(device,dest,message), which allows you to send an SMS message via the dialplan, thru the bluetooth attached phone.&lt;br /&gt;&lt;br /&gt;To get an SMS, you have to have a cellphone bluetooth attached, and capable of passing sms messages. When it reports to Asterisk via the bluetooth connection, that an SMS message was recieved, Asterisk will try to run the &quot;sms&quot; extension, with the channel variables &lt;em&gt;SMSSRC&lt;/em&gt; and &lt;em&gt;SMSTXT&lt;/em&gt; channel variables set to the appropriate values. ...</description>
            <author>sequille</author>
            <pubDate>Fri, 05 Aug 2011 15:18:21 +0100</pubDate>
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