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        <title>IG Manual sip</title>
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       <dc:date>2026-04-14T14:31:24+02:00</dc:date>
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                <rdf:li rdf:resource="http://wiki.igmanual.com/doku.php?id=sip:start&amp;rev=1315223478&amp;do=diff"/>
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        <title>IG Manual</title>
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    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:auto_attendant&amp;rev=1385454646&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2013-11-26T09:30:46+02:00</dc:date>
        <title>sip:auto_attendant</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:auto_attendant&amp;rev=1385454646&amp;do=diff</link>
        <description>The auto attendant is a SIP client which can be called like a usual SIP telephone. Just that you do not have to answer the calls yourself, instead the Internet gate does this for you, all you have to do is to configure how the auto attendant shall operate.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:certificates&amp;rev=1290160777&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-19T10:59:37+02:00</dc:date>
        <title>sip:certificates</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:certificates&amp;rev=1290160777&amp;do=diff</link>
        <description>By using certificates you can use SIP TLS (Transport Layer Security) encrypting your SIP traffic.
Certificates are used to verify the other end of a SIP connection is a known user, not an imposter.

The Internet Gate can use certificates created by other authorities, create its own certificates, or even create certificates for clients connecting to it.</description>
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    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:conference_rooms&amp;rev=1340703074&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2012-06-26T11:31:14+02:00</dc:date>
        <title>sip:conference_rooms</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:conference_rooms&amp;rev=1340703074&amp;do=diff</link>
        <description>This feature is available for releases &gt;= 5.34.

The Internet Gate provides built-in conference rooms for audio conferences.

There are up to two conference units, each one having its own sip adress, which can be called by someone who wants to dial into a conference.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:dial_plan&amp;rev=1305115233&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-05-11T14:00:33+02:00</dc:date>
        <title>sip:dial_plan</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:dial_plan&amp;rev=1305115233&amp;do=diff</link>
        <description>The Dial Plan is primarily intended for routing outbound calls if the Internet Gate is used as the SIP Server for the callers domain. If another SIP Server is used (the domain of the Request-URI) the call will automatically route to that server and the dial plan is not likely needed (however the Internet Gate can be configured to apply the dial plan to such calls as well).</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:enum&amp;rev=1290003292&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-17T15:14:52+02:00</dc:date>
        <title>sip:enum</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:enum&amp;rev=1290003292&amp;do=diff</link>
        <description>ENUM stands for E.164 NUmber Mapping and allows telephone calls that usually would have been made on the normal telephony network to be made over the Internet instead.

Read more about wp&gt;ENUM.

Internet Gate can easily be configured to use ENUM using the Dial Plan.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:fent&amp;rev=1289299668&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-09T11:47:48+02:00</dc:date>
        <title>sip:fent</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:fent&amp;rev=1289299668&amp;do=diff</link>
        <description>The Internet Gate can enable SIP connectivity for remote users that use NAT devices without SIP support. It can adapt to characteristics of remote NAT devices.

Using FENT you can allow people behind non-SIP-capable NAT firewalls to register on your Internet Gate's SIP server.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:listen_to_voicemail&amp;rev=1315569773&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-09-09T14:02:53+02:00</dc:date>
        <title>sip:listen_to_voicemail</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:listen_to_voicemail&amp;rev=1315569773&amp;do=diff</link>
        <description>The built-in voicemail unit can record messages when users are not answering their phones.

The recorded messages can either be e-mailed to the users' e-mail accounts, or kept in the USB memory stick attached to the Internet Gate for later listening using web browser or phone.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:locating_sip_services&amp;rev=1308657015&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-06-21T13:50:15+02:00</dc:date>
        <title>sip:locating_sip_services</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:locating_sip_services&amp;rev=1308657015&amp;do=diff</link>
        <description>It is very convenient to be able to use one's personal address, i.e. john@company.com, for both email and SIP communications (IP telephony, presence, video conferencing etc), just by changing the application prefix.

For example: email:john@company.com and sip:john@company.com</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:set_up_a_sip_server&amp;rev=1289293785&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-09T10:09:45+02:00</dc:date>
        <title>sip:set_up_a_sip_server</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:set_up_a_sip_server&amp;rev=1289293785&amp;do=diff</link>
        <description>The way the Internet Gate handles SIP traffic makes it useful for more than just a firewall that supports SIP traffic. You may for instance set up the Internet Gate to work as a small SIP server.

Set up a SIP Server for your Domain


You can set up the Internet Gate as a SIP server for your existing domain, making it part of the open SIP world. Communicate with anyone in the open SIP community for free. Use any SIP service like video, presence (client based), instant messaging, and data collabo…</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:sip_clients&amp;rev=1306912027&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-06-01T09:07:07+02:00</dc:date>
        <title>sip:sip_clients</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:sip_clients&amp;rev=1306912027&amp;do=diff</link>
        <description>There are a lot of SIP clients on the market to choose from, there are free soft clients, affordable soft phones, and hardware phones. Here is a list of available SIP client setup instructions:

SIP Soft Phone Clients:


Windows Messenger

X-Lite

EyeBeam</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:sip_configurations&amp;rev=1288701977&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-02T13:46:17+02:00</dc:date>
        <title>sip:sip_configurations</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:sip_configurations&amp;rev=1288701977&amp;do=diff</link>
        <description>The Internet Gate fully supports the Session Initiation Protocol (SIP). SIP is used to establish multimedia sessions between users on a network (i.e. the Internet).

By default the Internet Gate is completely SIP transparent. The Internet Gate analyses the passing SIP traffic and dynamically controls the firewall to make multimedia packages pass through.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:sip_pbx&amp;rev=1327394871&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2012-01-24T09:47:51+02:00</dc:date>
        <title>sip:sip_pbx</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:sip_pbx&amp;rev=1327394871&amp;do=diff</link>
        <description>SIP PBX requires firmware release 5.33 or later installed in the unit.

SIP PBX requires purchase of  to be activated.

SIP PBX requires a USB memory stick to be inserted into the unit's USB port for optimal functionality.

SIP Switch

The SIP switch is the main component of the Internet Gate's SIP PBX feature. The SIP switch routes incoming and outgoing calls depending on who called whom and at what time of the day, what extension numbers were called, authentications, blacklists, etc.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:sip_solutions&amp;rev=1288691109&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-02T10:45:09+02:00</dc:date>
        <title>sip:sip_solutions</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:sip_solutions&amp;rev=1288691109&amp;do=diff</link>
        <description>The Internet Gate fully supports the SIP protocol, which means that you may use SIP based services and applications all the way from the Internet, through the firewall, and on to your secure inside network.

The Internet Gate keeps track of all connected SIP users. It analyses all passing SIP traffic, handles NAT related issues with private IP addresses and routes incoming as well as outgoing messages to the correct destination/user.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:sip_switch&amp;rev=1290002611&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-17T15:03:31+02:00</dc:date>
        <title>sip:sip_switch</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:sip_switch&amp;rev=1290002611&amp;do=diff</link>
        <description>Your Internet Gate can be software upgraded with the SIP Switch software. This software can enhance the functionality of your Internet Gate and give it PBX-like features for software and hardware SIP clients integrating them with each other, the SIP world and the old telephony network via gateways.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:sip_user_scenarios&amp;rev=1296758743&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-02-03T19:45:43+02:00</dc:date>
        <title>sip:sip_user_scenarios</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:sip_user_scenarios&amp;rev=1296758743&amp;do=diff</link>
        <description>The Internet Gate fully supports the SIP protocol, which means that you may use SIP based services and applications all the way from the Internet, through the firewall, and on to your secure inside network.

The Internet Gate keeps a database of all connected SIP users. It analyses all passing SIP traffic and distributes it to the right user on the inside network.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:start&amp;rev=1315223478&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-09-05T13:51:18+02:00</dc:date>
        <title>sip:start</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:start&amp;rev=1315223478&amp;do=diff</link>
        <description>SIP (Session Initiation Protocol) is an Internet protocol for live communication between persons. The Internet Gate is equipped with a SIP proxy and registrar that dynamically controls the firewall to allow multiple SIP clients on the LAN to communicate universally over the Internet. With the Internet Gate you can communicate using telephony, presence, instant messaging, voice and video applications all the way into your private LAN and still being protected by a vigorous firewall.</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:start_using_sip&amp;rev=1288702607&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-02T13:56:47+02:00</dc:date>
        <title>sip:start_using_sip</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:start_using_sip&amp;rev=1288702607&amp;do=diff</link>
        <description>The fact that you own a firewall that is fully SIP transparent, enables you to start using the new exciting services based on SIP, all the way onto your private network.

If you are new to SIP, and want to test this new form of Internet communication, the easiest way is to download and install Microsoft® Windows Messenger. With Microsoft® Windows Messenger you can use SIP-based telephony, presence, instant messaging and voice/video communication to see when your friends are online, exchange inst…</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:telephony_ports&amp;rev=1301928829&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-04-04T16:53:49+02:00</dc:date>
        <title>sip:telephony_ports</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:telephony_ports&amp;rev=1301928829&amp;do=diff</link>
        <description></description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:test_agent&amp;rev=1327400155&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2012-01-24T11:15:55+02:00</dc:date>
        <title>sip:test_agent</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:test_agent&amp;rev=1327400155&amp;do=diff</link>
        <description>This feature is available for releases &gt;= 5.34. It does not require any licenses. It is not available if the Internet gate has a flash size less than 8MB.

In order for users to have an easy-to-use possibility to test the audio devices of their telephone or soft client, the Internet gate has two integrated SIP endpoints acting as Test Agents, which by default can be called either with the sip adress “testme” (or with Alias 8888) and “testservice” (or with Alias 8889).</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:troubleshooting&amp;rev=1289384305&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-10T11:18:25+02:00</dc:date>
        <title>sip:troubleshooting</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:troubleshooting&amp;rev=1289384305&amp;do=diff</link>
        <description>Phone cannot register

Turn off ICE, STUN, uPnP and other “tricks” that your SIP phone might try to use to get through ordinary firewalls. As the Internet Gate is SIP transparent such “tricks” are harmful and unnecessary - and might even actually stop SIP traffic from getting through the firewall!</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:trunk&amp;rev=1290085921&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-11-18T14:12:01+02:00</dc:date>
        <title>sip:trunk</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:trunk&amp;rev=1290085921&amp;do=diff</link>
        <description>The SIP Trunk module is designed to connect IP-PBX's and other SIP endpoints (phones) to Internet Telephony Service Providers by acting as a B2BUA (Back-To-Back User Agent) with one interface to the ITSP and the other to the IP-PBX and SIP phones. The ITSP interface is called SIP Trunk and the protocol used on both interfaces are SIP. The main function performed by this module is to provide a demarcation point where all communication on the SIP trunk behaves in a predictable way regardless of wh…</description>
    </item>
    <item rdf:about="http://wiki.igmanual.com/doku.php?id=sip:voicemail&amp;rev=1315307169&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-09-06T13:06:09+02:00</dc:date>
        <title>sip:voicemail</title>
        <link>http://wiki.igmanual.com/doku.php?id=sip:voicemail&amp;rev=1315307169&amp;do=diff</link>
        <description>The Internet Gate can record voice messages left by callers to a USB memory stick attached to the unit.

The recorded messages can then either be e-mailed to the called user, or she can listen to them using web browser or telephone.

Voice mail requires purchase of  to be activated.</description>
    </item>
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