[Asterisk-doc] docs glossary.xml,1.5,1.6 modules.xml,1.4,1.5
blitzrage
asterisk-doc@lists.digium.com
Sun, 15 Aug 2004 13:33:43 +0000
Comments:
Update of /cvsroot/asterisk/docs
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv18945/docs
Modified Files:
glossary.xml modules.xml
Log Message:
blitzrage
- minor corrections to the capi.conf section by Martin List-Petersen
- added a couple of terms to the glossary
Index: glossary.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/glossary.xml,v
retrieving revision 1.5
retrieving revision 1.6
diff -C2 -d -r1.5 -r1.6
*** glossary.xml 1 Jun 2004 20:40:18 -0000 1.5
--- glossary.xml 15 Aug 2004 13:33:37 -0000 1.6
***************
*** 1,25 ****
<glossary>
<title>Glossary of Asterisk & Telecom Terms</title>
! <glossentry>
<glossterm>FXO</glossterm>
<glossdef>
<para>Foreign eXchange Office</para>
<para>
! An FXO device can be an analog phone, answering machine, fax, or anything
! that handles a call from the telephone company like one. They should also
! operate the same way when connected to an FXS interface.
</para>
<para>
! An FXO interface will accept calls from FXS or PSTN interfaces. All countries
! and regions have their own standards.
</para>
<para>
! FXO is complimentary to FXS (and the PSTN).
</para>
</glossdef>
</glossentry>
! <glossentry>
<glossterm>FXS</glossterm>
<glossdef>
--- 1,25 ----
<glossary>
<title>Glossary of Asterisk & Telecom Terms</title>
! <glossentry sortas="FXO">
<glossterm>FXO</glossterm>
<glossdef>
<para>Foreign eXchange Office</para>
<para>
! An FXO device can be an analog phone, answering machine, fax, or anything
! that handles a call from the telephone company like one. They should also
! operate the same way when connected to an FXS interface.
</para>
<para>
! An FXO interface will accept calls from FXS or PSTN interfaces. All countries
! and regions have their own standards.
</para>
<para>
! FXO is complimentary to FXS (and the PSTN).
</para>
</glossdef>
</glossentry>
! <glossentry sortas="FXS">
<glossterm>FXS</glossterm>
<glossdef>
***************
*** 31,44 ****
<para>
! An FXS device will allow any FXO device to operate as if it were connected to
! the phone company. This makes your PBX the POTS+PSTN for the phone.
</para>
<para>
! The FXS Interface connects to FXO devices (by an FXO interface, of course).
</para>
</glossdef>
</glossentry>
! <glossentry>
<glossterm>PSTN</glossterm>
<glossdef>
--- 31,44 ----
<para>
! An FXS device will allow any FXO device to operate as if it were connected to
! the phone company. This makes your PBX the POTS+PSTN for the phone.
</para>
<para>
! The FXS Interface connects to FXO devices (by an FXO interface, of course).
</para>
</glossdef>
</glossentry>
! <glossentry sortas="PSTN">
<glossterm>PSTN</glossterm>
<glossdef>
***************
*** 49,53 ****
</glossdef>
</glossentry>
! <glossentry>
<glossterm>ADSI</glossterm>
<glossdef>
--- 49,53 ----
</glossdef>
</glossentry>
! <glossentry sortas="ADSI">
<glossterm>ADSI</glossterm>
<glossdef>
***************
*** 71,75 ****
</glossdef>
</glossentry>
! <glossentry>
<glossterm>ISDN</glossterm>
<glossdef>
--- 71,75 ----
</glossdef>
</glossentry>
! <glossentry sortas="ISDN">
<glossterm>ISDN</glossterm>
<glossdef>
***************
*** 82,86 ****
</glossdef>
</glossentry>
! <glossentry>
<glossterm>PRI</glossterm>
<glossdef>
--- 82,86 ----
</glossdef>
</glossentry>
! <glossentry sortas="PRI">
<glossterm>PRI</glossterm>
<glossdef>
***************
*** 93,97 ****
</glossdef>
</glossentry>
! <glossentry>
<glossterm>BRI</glossterm>
<glossdef>
--- 93,97 ----
</glossdef>
</glossentry>
! <glossentry sortas="BRI">
<glossterm>BRI</glossterm>
<glossdef>
***************
*** 103,107 ****
</glossdef>
</glossentry>
! <glossentry>
<glossterm>VoIP</glossterm>
<glossdef>
--- 103,107 ----
</glossdef>
</glossentry>
! <glossentry sortas="VoIP">
<glossterm>VoIP</glossterm>
<glossdef>
***************
*** 115,118 ****
</glossdef>
</glossentry>
!
</glossary>
--- 115,152 ----
</glossdef>
</glossentry>
! <glossentry sortas="MSN">
! <glossterm>MSN</glossterm>
! <glossdef>
! <para>Multiple Subscriber Line</para>
! <para>
! This is a telephone number associated with an ETS 300 BRI line. Providers of
! ETS 300 often give you three MSNs with a BRI, although additional MSNs can be
! purchased. An ISDN terminal will "ring" (provide an alerting signal) only when
! calls are made to the MSN (or MSNs) entered in that terminal. If a terminal has
! no MSNs entered it will "ring" whenever there is a call to any of the MSN.s on
! that BRI.
! </para>
! </glossdef>
! </glossentry>
! <glossentry sortas="DID">
! <glossterm>DID</glossterm>
! <glossdef>
! <para>Direct Inward Dialing</para>
! <para>
! Direct Inward Dialing. The ability for an outside caller to dial to a PBX
! extension without going through an attendant or auto-attendant.
! </para>
! </glossdef>
! </glossentry>
! <glossentry sortas="DTMF">
! <glossterm>DTMF</glossterm>
! <glossdef>
! <para>Dual Tone Multi Frequency</para>
! <para>
! The standard tone-pairs used on telephone terminals for dialing using in-band
! signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most
! terminals support only 12 of them (0-9, * and #).
! </para>
! </glossdef>
! </glossentry>
</glossary>
Index: modules.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/modules.xml,v
retrieving revision 1.4
retrieving revision 1.5
diff -C2 -d -r1.4 -r1.5
*** modules.xml 15 Aug 2004 05:08:15 -0000 1.4
--- modules.xml 15 Aug 2004 13:33:37 -0000 1.5
***************
*** 345,349 ****
<title>capi.conf</title>
<para>
! The configuration of capi based devices is pretty simple. First of all do vi have the <varname>[general]</varname>
section.
<informalexample>
--- 345,350 ----
<title>capi.conf</title>
<para>
! The configuration of capi based devices is fairly straight forward. The first
! section we need to edit is <varname>[general]</varname>.
section.
<informalexample>
***************
*** 356,370 ****
</programlisting>
</informalexample>
! <varname>nationalprefix</varname> and <varname>internationalprefix</varname> define, which prefix
! Asterisk should add in front of the caller-id on incoming calls. ISDN works that way, that the caller-id
! is transferred without prefix and there is a special indicator, that tells on call setup, what kind of
! call we are talking about. Since Asterisk doesn't differ and the prefix can be different from country to
! country (international is "011" in the U.S., but "00" in most european countries) this can be set here.
! <varname>rxgain</varname> and <varname>txgain</varname> do explain themselfes.
</para>
<para>
! Next up is the <varname>[interfaces]</varname> section. This section defines, which capi interfaces
! you want to add to Asterisk and which numbers are available. ISDN can be configured two ways, Point
! to Multi-Point (PMP) or Point to Point (PP). Here is a typical PMP setup:
<informalexample>
<programlisting>
--- 357,374 ----
</programlisting>
</informalexample>
! <varname>nationalprefix</varname> and <varname>internationalprefix</varname> define which prefix
! Asterisk should add in front of the caller-id on incoming calls. ISDN works like this; the caller-id
! is transferred without the prefix and there is a special indicator, on call setup, what kind of
! call we are talking about. Since Asterisk does not differenciate between different countries,
! and the prefix can be different from country to country, this can be set here. For example, the
! international prefix in North America is "011", but in most European countries it is "00".
! <varname>rxgain</varname> and <varname>txgain</varname> allow you to adjust the receiving and
! transmitting gain (volume) respectfully. These numbers are in decibals (dB) so only adjust in
! small incremements. Setting these too high will cause Asterisk to produce echo on the channel.
</para>
<para>
! Next up is the <varname>[interfaces]</varname> section. This section defines which capi interfaces
! you want to add to Asterisk and which numbers are available. ISDN can be configured two ways: Point
! to Multi-Point (PMP) or Point to Point (PtP). Here is a typical PMP setup:
<informalexample>
<programlisting>
***************
*** 378,382 ****
context=default
callgroup=1
! ;mode=immidiate
;deflect=12345678
;echosquelch=1
--- 382,386 ----
context=default
callgroup=1
! ;mode=immediate
;deflect=12345678
;echosquelch=1
***************
*** 385,393 ****
</programlisting>
</informalexample>
! <varname>msn</varname> specifies which MSNs (or DIDs) you want to use for outgoing calls. There is
! a limitation of up to 5 MSNs, but usually it is enough to specify one of them. You can set the MSN
! you want to show on outgoing calls by setting the <varname>${CALLERIDNUM}</varname> variable in your
! dialplan. On <varname>incomingmsn</varname> you can specify, which MSNs Asterisk should react on. A
! "*" tells Asterisk to react on any call, no matter what MSN it has.
</para>
<para>
--- 389,398 ----
</programlisting>
</informalexample>
! <varname>msn</varname> specifies which <glossterm>MSN</glossterm>s (or <glossterm>DID</glossterm>s)
! you want to use for outgoing calls. There is a limitation of up to 5 MSNs, but usually it is enough
! to specify one of them. You can set the MSN you want to show on outgoing calls by setting the
! <varname>${CALLERIDNUM}</varname> variable in your dialplan. On <varname>incomingmsn</varname>
! you can specify, which MSNs Asterisk should react on. A "*" tells Asterisk to react on any call,
! no matter what MSN it has.
</para>
<para>
***************
*** 395,422 ****
than one controller in your system and all can be configured individually. <varname>devices</varname>
tells chan_capi how many b-channels your ISDN card can handle. <varname>softdtmf=1</varname> will use
! Asterisk's dsp functions to detect and generate the DTMF tones. <varname>softdtmf=0</varname> will
! use your capi controller to do the detection/generation. Unless you have an active card you should use
! <varname>softdtmf=1</varname>.
</para>
<para>
! <varname>accountcode</varname> is for billing purposes and <varname>context</varname> defines which
! context chan_capi should send incomming calls from the ISDN card to. <varname>callgroup</varname>
defines which callgroup the controller is a member of. You can have several controllers in the same
! callgroup, that then would react the same way. If <varname>mode</varname> is set to immidiate, the
! ISDN card answers the call immidiatly and passes it on to Asterisk. The default behavior would be,
that you need to answer the call in your dialplan.
</para>
<para>
! <varname>deflect</varname> defines a phoneno. that you automatically want to deflect calls to, if
both b-channels on your ISDN card are busy. Deflect on busy has to be enabled during compile,
if you want that feature to work.
</para>
<para>
<varname>echosquelch</varname>, <varname>echocancel</varname> and <varname>echotail</varname> are
! values, that you normally not need to change. If you are dealing with echo issues, it might be an
! idea to try test different values here.
</para>
<para>
! The typical PP setup would look like this:
<informalexample>
<programlisting>
--- 400,428 ----
than one controller in your system and all can be configured individually. <varname>devices</varname>
tells chan_capi how many b-channels your ISDN card can handle. <varname>softdtmf=1</varname> will use
! Asterisk's dsp functions to detect and generate the <glossterm>DTMF</glossterm> tones.
! <varname>softdtmf=0</varname> will use your capi controller to do the detection/generation. Unless
! you have an active card you should use <varname>softdtmf=1</varname>.
</para>
<para>
! <varname>accountcode</varname> is for billing purposes. <varname>context</varname> defines which
! context chan_capi should send incoming calls from the ISDN card to. <varname>callgroup</varname>
defines which callgroup the controller is a member of. You can have several controllers in the same
! callgroup, that then would react in the same way. If <varname>mode</varname> is set to immediate, the
! ISDN card answers the call and immediatly passes it on to Asterisk. The default behavior would be,
that you need to answer the call in your dialplan.
</para>
<para>
! <varname>deflect</varname> defines a phone number that you automatically want to deflect calls to if
both b-channels on your ISDN card are busy. Deflect on busy has to be enabled during compile,
if you want that feature to work.
+ <!-- Lets explain somewhere how to enable this -->
</para>
<para>
<varname>echosquelch</varname>, <varname>echocancel</varname> and <varname>echotail</varname> are
! values, that you normally need not to change. If you are dealing with echo issues, it might be an
! idea to try testing different values here.
</para>
<para>
! The typical PtP setup would look like this:
<informalexample>
<programlisting>
***************
*** 433,437 ****
</informalexample>
The options for a Point to Point setup are basically the same as for the PMP setup. <varname>msn</varname>
! now only holds the prefix of your msn, the suffix will be transferred to Asterisk as extension.
<varname>isdnmode=ptp</varname> tells chan_capi, that the card is to be run in Point to Point mode (Default
would be Point to Multi-Point mode). All other values are to be configured the same way with Point to
--- 439,443 ----
</informalexample>
The options for a Point to Point setup are basically the same as for the PMP setup. <varname>msn</varname>
! now only holds the prefix of your msn, the suffix will be transferred to Asterisk as an extension.
<varname>isdnmode=ptp</varname> tells chan_capi, that the card is to be run in Point to Point mode (Default
would be Point to Multi-Point mode). All other values are to be configured the same way with Point to