[Asterisk-doc] docs installation.xml,1.13,1.14 voip-providers.xml,1.2,1.3
blitzrage
asterisk-doc@lists.digium.com
Wed, 2 Jun 2004 00:24:49 +0000
Comments:
Update of /cvsroot/asterisk/docs
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv27016/docs
Modified Files:
installation.xml voip-providers.xml
Log Message:
Martin List-Petersen
- installation.xml patch
- voip-providers.xml patch (to eventually be moved to website)
Index: installation.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/installation.xml,v
retrieving revision 1.13
retrieving revision 1.14
diff -C2 -d -r1.13 -r1.14
*** installation.xml 31 May 2004 02:06:33 -0000 1.13
--- installation.xml 2 Jun 2004 00:24:43 -0000 1.14
***************
*** 74,82 ****
<title>IRQ Sharing Issues</title>
<para>
! Many telephony cards such as the X100P tend to generate a large amount of interupts.
Servicing them takes time and drivers may not be able to do it on-time. If another device is
! processing the same shared IRQ, the IRQ line connot receive another one. It does
tend to work better on SMP (APIC) systems. On single chip systems you can get
! interupt misses and messed up clocking. Any of Digium's cards or other telephony
cards can be subject to this problem. Because the precise delivery of IRQ's is
very much necessary in telephony, one should not share IRQ's with anything.
--- 74,82 ----
<title>IRQ Sharing Issues</title>
<para>
! Many telephony cards such as the X100P tend to generate a large amount of interrupts.
Servicing them takes time and drivers may not be able to do it on-time. If another device is
! processing the same shared IRQ, the IRQ line cannot receive another one. It does
tend to work better on SMP (APIC) systems. On single chip systems you can get
! interrupt misses and messed up clocking. Any of Digium's cards or other telephony
cards can be subject to this problem. Because the precise delivery of IRQ's is
very much necessary in telephony, one should not share IRQ's with anything.
***************
*** 122,166 ****
<para>[This should probably be moved somewhere else.]</para>
<para>
! ISDN hardware in most parts of the world are not very expensive. A basic AVM card
! that comes with CAPI compatible kernel modules is available for about $40US.
! Sometimes there are several differences between the capacity of the cards such as
! having more than 2 B-channels and supporting different ISDN standards. The chan_capi
! module is programmed to work even with multiple ISDN cards. To use chan_capi you must
! have support for CAPI and your ISDN card in your kernel configuration.
! You will have to check with your vendor to see if your particular card is supported, but
! Linux has good support for ISDN cards, so most likely it will. Cards such as Eicon and AVM
! are known to work with Asterisk. However it can be a bit tricky to get CAPI
! support working for different ISDN cards. The following links have good
! descriptions on how to get CAPI support for several cards.
! </para>
! <para>
! For a "Eicon Diva" visit:
! </para>
! <para>
! <simplelist>
! <member><ulink url="http://www.melware.de/de/index.html" type="http">http://www.melware.de/de/index.html</ulink></member>
! <member><ulink url="http://isdn4linux.org/~armin/divas/" type="http">http://isdn4linux.org/~armin/divas/</ulink></member>
! <member><ulink url="http://www.eicon.com/worldwide/products/WAN/cn4linux.htm" type="http">http://www.eicon.com/worldwide/products/WAN/cn4linux.htm</ulink></member>
! </simplelist>
! </para>
! <para>
! For AVM cards visit:
! </para>
! <para>
! <ulink url="http://www.avm.de" type="http">http://www.avm.de</ulink>
! </para>
! <para>
! For AVM-Fritz! cards:
! </para>
! <para>
! <ulink url="ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/" type="http">ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/</ulink> (not only for SuSE)
! <ulink url="http://www.linux-magazin.de/Artikel/ausgabe/2000/10/Capi/capi.html" type="http">http://www.linux-magazin.de/Artikel/ausgabe/2000/10/Capi/capi.html</ulink>
! </para>
</sect2>
--- 122,211 ----
<para>[This should probably be moved somewhere else.]</para>
<para>
! ISDN hardware in most parts of the world are not very expensive. A basic AVM card
! that comes with CAPI compatible kernel modules is available for about $40US.
! Sometimes there are several differences between the capacity of the cards such as
! having more than 2 B-channels and supporting different ISDN standards.
! </para>
! <sect3>
! <title>chan_capi</title>
! <para>
! This channel driver is programmed to work even with multiple ISDN cards.
! To use chan_capi you must have support for CAPI and your ISDN card in your
! kernel configuration. You will have to check with your vendor to see if your
! particular card is supported, but Linux has good support for ISDN cards, so
! most likely it will. Cards such as Eicon and AVM are known to work with
! Asterisk. However it can be a bit tricky to get CAPI support working for
! different ISDN cards. Be also aware, that chan_capi only provides TE mode,
! you can only use it for connecting your asterisk box to a ISDN line, not connect
! phones internally to the ISDN card. The following links have good descriptions
! on how to get CAPI support for several cards.
! </para>
! <para>
! chan_capi is a third party channel driver, that can be downloaded at <ulink url="http://www.junghanns.net/asterisk/" type="http">http://www.junghanns.net/asterisk/</ulink>
! </para>
!
! <para>
! For a "Eicon Diva" visit:
! </para>
! <para>
! <simplelist>
! <member><ulink url="http://www.melware.de/de/index.html" type="http">http://www.melware.de/de/index.html</ulink></member>
! <member><ulink url="http://isdn4linux.org/~armin/divas/" type="http">http://isdn4linux.org/~armin/divas/</ulink></member>
! <member><ulink url="http://www.eicon.com/worldwide/products/WAN/cn4linux.htm" type="http">http://www.eicon.com/worldwide/products/WAN/cn4linux.htm</ulink></member>
! </simplelist>
! </para>
! <para>
! For AVM cards visit:
! </para>
! <para>
! <ulink url="http://www.avm.de" type="http">http://www.avm.de</ulink>
! </para>
! <para>
! For AVM-Fritz! cards:
! </para>
! <para>
! <ulink url="ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/" type="http">ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/</ulink> (not only for SuSE)
! <ulink url="http://www.linux-magazin.de/Artikel/ausgabe/2000/10/Capi/capi.html" type="http">http://www.linux-magazin.de/Artikel/ausgabe/2000/10/Capi/capi.html</ulink> (German article on getting capi to work with AVM Fritz!)
! </para>
! </sect3>
! <sect3>
! <title>zapbri, qozap</title>
! <para>
! The zapbri and qozap drivers take a bit different approach than chan_capi: they implement a full zaptel driver
! for asterisk on ISDN cards based on the HFC chipset and can provide either TE and NT mode on a per port basis.
! As with all zaptel drivers you can have any amount of these cards in your asterisk box.
! </para>
!
! <para>
! The qozap driver is for 4 port (quadBRI) and 8 port (octoBRI) ISDN cards that also are designed and sold by it's
! developer. These cards can provide power and termination on the port for NT mode and do offer an internal PCM bus,
! that connects all quadBRI or octoBRI cards together to interconnect up to 64 full duplex connections at 64 kbit/s each
! without using that bandwidth on the PCI bus or the host CPU.
! </para>
!
! <para>
! The zaphfc driver is basically the same driver just for a single port ISDN card available at about 30$ in most hardware
! stores. TE mode, hooking it up against your ISDN line, is trivial and works as good as with the capi based cards,
! however to get NT mode to work with this card you will need to build yourself an ISDN crossover cable and provide power
! and termination on the ISDN bus. This can be done by taking an old NT1 that not is in use anymore and modify it a little.
! </para>
!
! <para>
! These drivers are quite new and require to patch both asterisk, libpri and build the kernel module (zapbri/qozap) to
! get them working, but once this is done you have full featured zap channels on these cards, which makes them really interesting.
! </para>
!
! <para>
! zapbri and qozap are third party modules for asterisk and can be downloaded at:
! <ulink url="http://www.junghanns.net/asterisk/" type="http">http://www.junghanns.net/asterisk/</ulink>. The package to download
! there is called "bristuff".
! </para>
! </sect3>
</sect2>
Index: voip-providers.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/voip-providers.xml,v
retrieving revision 1.2
retrieving revision 1.3
diff -C2 -d -r1.2 -r1.3
*** voip-providers.xml 11 Jan 2004 08:09:33 -0000 1.2
--- voip-providers.xml 2 Jun 2004 00:24:43 -0000 1.3
***************
*** 9,33 ****
<title>Free Service Providers</title>
<sect2>
! <title>IAXTEL</title>
<para/>
<sect3>
<title>Description</title>
! <para/>
</sect3>
<sect3>
<title>Services</title>
! <para/>
</sect3>
<sect3>
<title>What to Expect</title>
! <para/>
! </sect3>
! <sect3>
! <title>Hardware</title>
! <para/>
</sect3>
<sect3>
<title>Setup Examples</title>
! <para/>
</sect3>
<sect3>
--- 9,106 ----
<title>Free Service Providers</title>
<sect2>
! <title><ulink url="http://www.fwdnet.net/" type="url">Free World Dialup (FWD)</ulink></title>
<para/>
<sect3>
<title>Description</title>
! <para>
! Free World Dialup is a free SIP Directory Service.
! </para>
</sect3>
<sect3>
<title>Services</title>
! <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>
! 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>
! 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>
***************
*** 36,73 ****
<sect4>
<title>Protocols</title>
! <para/>
! </sect4>
! <sect4>
! <title>Troubleshooting</title>
<para/>
</sect4>
<sect4>
<title>Help</title>
! <para/>
</sect4>
</sect3>
</sect2>
<sect2>
! <title>FWD</title>
<para/>
<sect3>
<title>Description</title>
! <para/>
</sect3>
<sect3>
<title>Services</title>
! <para/>
! </sect3>
<sect3>
<title>What to Expect</title>
! <para/>
</sect3>
<sect3>
! <title>Hardware</title>
<para/>
</sect3>
<sect3>
<title>Setup Examples</title>
! <para/>
</sect3>
<sect3>
--- 109,263 ----
<sect4>
<title>Protocols</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>
! 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><ulink url="http://www.iaxtel.com/" type="url">IAXTEL</ulink></title>
<para/>
<sect3>
<title>Description</title>
! <para>
! IAXTel is a free IAX directory service provided by Digium, the company behind Asterisk.
! </para>
</sect3>
<sect3>
<title>Services</title>
! <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>
! 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>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/${EXTEN}@iaxtel)
! exten => _17XXNXXXXXX.,2,Hangup
! </programlisting>
! </example>
! </para>
! </sect3>
! <sect3>
! <title>Technical Setup</title>
<para/>
+ <sect4>
+ <title>Protocols</title>
+ <para>
+ IAXTel only supports IAX.
+ </para>
+ </sect4>
+ <sect4>
+ <title>Help</title>
+ <para>
+ Normally you won't need any help if you follow the examples on the website (or here). It is straight forward.
+ </para>
+ </sect4>
+ </sect3>
+ </sect2>
+ <sect2>
+ <title><ulink url="http://iptel.org/" type="url">IPTel</ulink></title>
+ <para/>
+ <sect3>
+ <title>Description</title>
+ <para>
+ IPTel is a free SIP Directory Service provided by the company that
+ develops SER (SIP Express Router).
+ </para>
+ </sect3>
+ <sect3>
+ <title>Services</title>
+ <para>
+ <orderedlist>
+ <listitem>
+ <para>Forwarding to another SIP URI.</para>
+ </listitem>
+ <listitem>
+ <para>Instant Messaging through SIP.</para>
+ </listitem>
+ </orderedlist>
+ </para>
</sect3>
<sect3>
<title>Setup Examples</title>
! <para>
! <example>
! <title>sip.conf</title>
! <programlisting>
! register => username:password@iptel.org
!
! [iptel]
! type=friend
! username=username
! secret=password
! host=iptel.org
! </programlisting>
! </example>
! <example>
! <title>extensions.conf</title>
! <programlisting>
! exten => _**478.,1,SetCallerID(Your Name <3400001234>)
! exten => _**478.,2,Dial(SIP/${EXTEN:5}@iptel)
! exten => _**478.,3,Hangup
! </programlisting>
! </example>
! </para>
</sect3>
<sect3>
***************
*** 76,113 ****
<sect4>
<title>Protocols</title>
! <para/>
! </sect4>
! <sect4>
! <title>Troubleshooting</title>
! <para/>
</sect4>
<sect4>
<title>Help</title>
! <para/>
</sect4>
</sect3>
</sect2>
<sect2>
! <title>SipPhone.com</title>
<para/>
<sect3>
<title>Description</title>
! <para/>
</sect3>
<sect3>
<title>Services</title>
! <para/>
</sect3>
<sect3>
<title>What to Expect</title>
! <para/>
</sect3>
<sect3>
! <title>Hardware</title>
<para/>
</sect3>
<sect3>
<title>Setup Examples</title>
! <para/>
</sect3>
<sect3>
--- 266,435 ----
<sect4>
<title>Protocols</title>
! <para>
! IPTel only supports SIP.
! </para>
</sect4>
<sect4>
<title>Help</title>
! <para>
! A lot tutorials etc., mostly concerning the products of iptel (SER).
! </para>
</sect4>
</sect3>
</sect2>
<sect2>
! <title><ulink url="http://www.sipgate.de/" type="url">SipGate</ulink></title>
<para/>
<sect3>
<title>Description</title>
! <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>
! <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>
! 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>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>
! <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>
***************
*** 116,128 ****
<sect4>
<title>Protocols</title>
! <para/>
! </sect4>
! <sect4>
! <title>Troubleshooting</title>
! <para/>
</sect4>
<sect4>
<title>Help</title>
! <para/>
</sect4>
</sect3>
--- 438,450 ----
<sect4>
<title>Protocols</title>
! <para>
! SIPPhone.com only supports SIP.
! </para>
</sect4>
<sect4>
<title>Help</title>
! <para>
! Support is mainly done through the forum on SIPPhone.com's website.
! </para>
</sect4>
</sect3>