[Asterisk-doc] voip-providers
Martin List-Petersen
asterisk-doc@lists.digium.com
Mon, 31 May 2004 23:34:51 +0100
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Here's sample setup and discription of some of the freely available
providers.
Comments on improvement are appreciated.
Kind regards,
Martin List-Petersen
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--- docs-current/voip-providers.xml 2004-05-31 13:12:54.000000000 +0100
+++ docs/voip-providers.xml 2004-05-31 23:30:33.000000000 +0100
@@ -8,122 +8,444 @@
<sect1>
<title>Free Service Providers</title>
<sect2>
- <title>IAXTEL</title>
+ <title><ulink url="http://www.fwdnet.net/" type="url">Free World Dialup (FWD)</ulink></title>
<para/>
<sect3>
<title>Description</title>
- <para/>
+ <para>
+ Free World Dialup is a free SIP Directory Service.
+ </para>
</sect3>
<sect3>
<title>Services</title>
- <para/>
+ <para>
+ <orderedlist>
+ <listitem>
+ <para>Voicemail</para>
+ </listitem>
+ <listitem>
+ <para>Toll free access to the US, UK, Netherlands and France</para>
+ </listitem>
+ <listitem>
+ <para>Conferencing</para>
+ </listitem>
+ <listitem>
+ <para>Free subscription to a direct phoneno. in the US (Washington, IPKall) and UK (CallUK)</para>
+ </listitem>
+ <listitem>
+ <para>
+ PSTN access numbers in Germany, the Netherlands, UK and different states of the U.S. enables
+ you to call any FWD subscriber from any regular phone.
+ </para>
+ </listitem>
+ </orderedlist>
+ This is only a limited list of what they provide you with. FWD also interconnects with a couple
+ of other directory services like iConnectHere, IAXTEL, Intertex, IPTel and NIC.at
+ </para>
</sect3>
<sect3>
<title>What to Expect</title>
- <para/>
- </sect3>
- <sect3>
- <title>Hardware</title>
- <para/>
+ <para>
+ FWD is in general very stable and developing their services all the time, remember however that FWD
+ not is a commercial service. Also they don't offer you any possibility to call PSTN no's beyond the
+ access to toll free numbers.
+ </para>
</sect3>
<sect3>
<title>Setup Examples</title>
- <para/>
+ <para>
+ First the way, how you can connect Asterisk to FWD by using SIP. This is the common setup used, since
+ the IAX gateway is fairly new and considered beta.
+ <example>
+ <title>sip.conf</title>
+ <programlisting>
+ register => 12345:password@fwd.pulver.com
+
+ [fwdnet]
+ type=friend
+ secret=password
+ host=fwd.pulver.com
+ dtmf=inband
+ allow=ulaw
+ </programlisting>
+ </example>
+ <example>
+ <title>extensions.conf</title>
+ <programlisting>
+ exten => _**393.,1,SetCallerID(Your Name <12345>)
+ exten => _**393.,2,Dial(SIP/${EXTEN:5}@fwdnet)
+ exten => _**393.,3,Hangup
+ </programlisting>
+ </example>
+ If you want to connect to FWD via IAX instead, you can subscribe for that in your profile. Your
+ configuration should look like this:
+ <example>
+ <title>iax.conf</title>
+ <programlisting>
+ register => 12345:password@iax2.fwdnet.net
+
+ [fwdnet]
+ type=user
+ allow=ulaw
+ deny=0.0.0.0/0.0.0.0
+ permit=65.39.205.0/255.255.255.0
+ </programlisting>
+ </example>
+ <example>
+ <title>extensions.conf</title>
+ <programlisting>
+ exten => _**393.,1,SetCallerID(Your Name <12345>)
+ exten => _**393.,2,Dial(IAX2/12345:password@iax2.fwdnet.net/${EXTEN:5})
+ exten => _**393.,3,Hangup
+ </programlisting>
+ </example>
+ The reason why ulaw is enabled is, that FWD's own services (voicemail, gateway to toll free numbers etc.
+ only support ulaw. FWD however passes all codecs through, if the subscriber you are calling supports these.
+ </para>
</sect3>
<sect3>
<title>Technical Setup</title>
<para/>
<sect4>
<title>Protocols</title>
- <para/>
- </sect4>
- <sect4>
- <title>Troubleshooting</title>
+ <orderedlist>
+ <listitem>
+ <para>SIP (fwd.pulver.com)</para>
+ </listitem>
+ <listitem>
+ <para>IAX (iax2.fwdnet.org)</para>
+ </listitem>
+ </orderedlist>
<para/>
</sect4>
<sect4>
<title>Help</title>
- <para/>
+ <para>
+ There are various ways of obtaining help with FWD: 55555 is the volunteer welcome line, 514 is the Virtual
+ Coffee House, 611 is the Technical Helpline (limited availability) or just use the user forums on their
+ website, which also can provide you with a lot helpful information.
+ </para>
</sect4>
</sect3>
</sect2>
<sect2>
- <title>FWD</title>
+ <title><ulink url="http://www.iaxtel.com/" type="url">IAXTEL</ulink></title>
<para/>
<sect3>
<title>Description</title>
- <para/>
+ <para>
+ IAXTel is a free IAX directory service provided by Digium, the company behind Asterisk.
+ </para>
</sect3>
<sect3>
<title>Services</title>
- <para/>
- </sect3>
+ <para>
+ <orderedlist>
+ <listitem>
+ <para>Toll free access to the US and Netherlands</para>
+ </listitem>
+ <listitem>
+ <para>
+ PSTN access number in U.S. enables you to call any IAXTel subscriber from any regular phone.
+ </para>
+ </listitem>
+ <listitem>
+ <para>
+ Interconnect numbers to FWD and VoicePulse subscribers.
+ </para>
+ </listitem>
+ </orderedlist>
+ </para>
+ </sect3>
<sect3>
<title>What to Expect</title>
- <para/>
+ <para>
+ IAXTel is in general quite stable, but don't get confused, when you look at the web interface. It is
+ quite outdated. Once you've subscribed to IAXTel you won't need the web interface anyway anymore.
+ </para>
</sect3>
<sect3>
- <title>Hardware</title>
+ <title>Setup Examples</title>
+ <para>
+ <example>
+ <title>iax.conf</title>
+ <programlisting>
+ register => username:password@iaxtel.com
+
+ [iaxtel]
+ type=user
+ auth=rsa
+ inkeys=iaxtel
+
+ [iaxtel2]
+ ;
+ ; Backwards compatible entry for IAXTel pre-RSA
+ ;
+ type=user
+ deny=0.0.0.0/0.0.0.0
+ permit=216.207.245.47/255.255.255.255
+ </programlisting>
+ </example>
+ <example>
+ <title>extensions.conf</title>
+ <programlisting>
+ exten => _17XXNXXXXXX.,1,Dial(IAX2/username:password@iaxtel.com/${EXTEN}@iaxtel)
+ exten => _17XXNXXXXXX.,2,Hangup
+ </programlisting>
+ </example>
+ </para>
+ </sect3>
+ <sect3>
+ <title>Technical Setup</title>
<para/>
+ <sect4>
+ <title>Protocols</title>
+ <para>
+ IAXTel only supports IAX.
+ </para>
+ </sect4>
+ <sect4>
+ <title>Help</title>
+ <para>
+ Normally you won't need any help if you follow the examples on the website (or here). It is straight forward.
+ </para>
+ </sect4>
+ </sect3>
+ </sect2>
+ <sect2>
+ <title><ulink url="http://iptel.org/" type="url">IPTel</ulink></title>
+ <para/>
+ <sect3>
+ <title>Description</title>
+ <para>
+ IPTel is a free SIP Directory Service provided by the company that
+ develops SER (SIP Express Router).
+ </para>
+ </sect3>
+ <sect3>
+ <title>Services</title>
+ <para>
+ <orderedlist>
+ <listitem>
+ <para>Forwarding to another SIP URI.</para>
+ </listitem>
+ <listitem>
+ <para>Instant Messaging through SIP.</para>
+ </listitem>
+ </orderedlist>
+ </para>
</sect3>
<sect3>
<title>Setup Examples</title>
- <para/>
+ <para>
+ <example>
+ <title>sip.conf</title>
+ <programlisting>
+ register => username:password@iptel.org
+
+ [iptel]
+ type=friend
+ username=username
+ secret=password
+ host=iptel.org
+ </programlisting>
+ </example>
+ <example>
+ <title>extensions.conf</title>
+ <programlisting>
+ exten => _**478.,1,SetCallerID(Your Name <3400001234>)
+ exten => _**478.,2,Dial(SIP/${EXTEN:5}@iptel)
+ exten => _**478.,3,Hangup
+ </programlisting>
+ </example>
+ </para>
</sect3>
<sect3>
<title>Technical Setup</title>
<para/>
<sect4>
<title>Protocols</title>
- <para/>
- </sect4>
- <sect4>
- <title>Troubleshooting</title>
- <para/>
+ <para>
+ IPTel only supports SIP.
+ </para>
</sect4>
<sect4>
<title>Help</title>
- <para/>
+ <para>
+ A lot tutorials etc., mostly concerning the products of iptel (SER).
+ </para>
</sect4>
</sect3>
</sect2>
<sect2>
- <title>SipPhone.com</title>
+ <title><ulink url="http://www.sipgate.de/" type="url">SipGate</ulink></title>
<para/>
<sect3>
<title>Description</title>
- <para/>
+ <para>
+ SIPGate is a German commercial SIP Directory Service, that gives you a free
+ direct phone number in Germany or the UK and access to toll free numbers in these countries.
+ </para>
</sect3>
<sect3>
<title>Services</title>
- <para/>
+ <para>
+ <orderedlist>
+ <listitem>
+ <para>Direct phone number in any of the cities in Germany or the UK, where SIPGate has a Gateway.</para>
+ </listitem>
+ <listitem>
+ <para>Toll free access to Germany and the UK.</para>
+ </listitem>
+ <listitem>
+ <para>Call forwarding to an IPTel SIP account, if you have one.</para>
+ </listitem>
+ <listitem>
+ <para>Online Call details of any incoming, outgoing or missed calls.</para>
+ </listitem>
+ <listitem>
+ <para>
+ Prepaid purchase of outgoing calls to any PSTN number, at quite competitive rates.
+ </para>
+ </listitem>
+ <listitem>
+ <para>
+ Free calls to other directory services like FWD, IPTel, SipPhone, IAXTel and Telio.no.
+ </para>
+ </listitem>
+ </orderedlist>
+ </para>
</sect3>
<sect3>
<title>What to Expect</title>
- <para/>
+ <para>
+ SIPGate is a German commercial provider, so their website is completely in German. If you can life with that,
+ you get a great service. They have been rated as one of the best VoIP providers available in Germany by most magazines.
+ </para>
</sect3>
<sect3>
- <title>Hardware</title>
+ <title>Setup Examples</title>
+ <para>
+ <example>
+ <title>sip.conf</title>
+ <programlisting>
+ register => 1234567:password@sipgate.de
+
+ [sipgate]
+ type=peer
+ secret=password
+ username=1234567
+ host=sipgate.de
+ </programlisting>
+ </example>
+ <example>
+ <title>extensions.conf</title>
+ <programlisting>
+ ;
+ ; 0 is needed in front of the area code for German phoneno.'s
+ ;
+ exten => _49ZX.,1,SetCallerID(Your Name <1234567>)
+ exten => _49ZX.,2,Dial(SIP/0${EXTEN:2}@sipgate)
+ exten => _49ZX.,3,Hangup
+
+ ;
+ ; 00 is the international dial code used in Europe.
+ ;
+ exten => _XXXX.,1,SetCallerID(Your Name <1234567>)
+ exten => _XXXX.,2,Dial(SIP/00${EXTEN}@sipgate)
+ exten => _XXXX.,3,Hangup
+ </programlisting>
+ </example>
+ </para>
+ </sect3>
+ <sect3>
+ <title>Technical Setup</title>
<para/>
+ <sect4>
+ <title>Protocols</title>
+ <para>
+ SIPGate only supports SIP.
+ </para>
+ </sect4>
+ <sect4>
+ <title>Help</title>
+ <para>
+ SIPGate has a great deal of online documentation in their "Help-Center". Basically everything you need, however German only.
+ </para>
+ </sect4>
+ </sect3>
+ </sect2>
+ <sect2>
+ <title><ulink url="http://www.sipphone.com/" type="url">SIPPhone.com</ulink></title>
+ <para/>
+ <sect3>
+ <title>Description</title>
+ <para>
+ SIPPhone.com is a free SIP Directory Service that originally only wanted to support hardphones,
+ now they also offer access with Xten.
+ </para>
+ </sect3>
+ <sect3>
+ <title>Services</title>
+ <para>
+ <orderedlist>
+ <listitem>
+ <para>Voicemail.</para>
+ </listitem>
+ <listitem>
+ <para>Conferencing.</para>
+ </listitem>
+ <listitem>
+ <para>
+ 5 calls of 1 minute each every day for free to Australia, Canada, France,
+ Germany, Hong Kong, Italy, Singapore, Taiwan, UK and the U.S.
+ </para>
+ </listitem>
+ <listitem>
+ <para>For an additional fee you can get a U.S. direct phoneno.</para>
+ </listitem>
+ </orderedlist>
+ </para>
</sect3>
<sect3>
<title>Setup Examples</title>
- <para/>
+ <para>
+ <example>
+ <title>sip.conf</title>
+ <programlisting>
+ register => 17471234567:password@proxy01.sipphone.com
+
+ [sipphone]
+ type=friend
+ secret=password
+ username=17471234567
+ host=proxy01.sipphone.com
+ </programlisting>
+ </example>
+ <example>
+ <title>extensions.conf</title>
+ <programlisting>
+ exten => _1747XXXXXXX,1,SetCallerID(Your Name <17471234567>)
+ exten => _1747XXXXXXX,2,Dial(SIP/${EXTEN}@sipphone)
+ exten => _1747XXXXXXX,3,Hangup
+ </programlisting>
+ </example>
+ </para>
</sect3>
<sect3>
<title>Technical Setup</title>
<para/>
<sect4>
<title>Protocols</title>
- <para/>
- </sect4>
- <sect4>
- <title>Troubleshooting</title>
- <para/>
+ <para>
+ SIPPhone.com only supports SIP.
+ </para>
</sect4>
<sect4>
<title>Help</title>
- <para/>
+ <para>
+ Support is mainly done through the forum on SIPPhone.com's website.
+ </para>
</sect4>
</sect3>
</sect2>
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