AsteriskGuru Archives

cvs at sohoskyway.net (ji · Guest

Comments:
Update of /cvsroot/asterisk/docs/volume-one
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv24004/volume-one

Modified Files:
vm1chp4-channelconfig.xml
Log Message:
minor typo correction
Index: vm1chp4-channelconfig.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/volume-one/vm1chp4-channelconfig.xml,v
retrieving revision 1.13
retrieving revision 1.14
diff -C2 -d -r1.13 -r1.14
*** vm1chp4-channelconfig.xml 10 Nov 2004 22:41:31 -0000 1.13
--- vm1chp4-channelconfig.xml 1 Feb 2005 21:49:40 -0000 1.14
***************
*** 148,152 ****
</informalexample>
<para>
! In the first example we listed channel 1 as an FXS channel. To create an FXO channel on the same TDM400P card, we list all the settings for the channel and then define the channel number. Instead of only having signalling be fxo_ks though we want the signalling to be fxs_ks. Because the other settings haven't been changed (signalling=fxs_ks replaces the previous value of signalling), they stay the same. Which means that while channel 1 takes on the values language=en, context=default, switchtype=national, and signalling=fxo_ks; channel takes on the values language=en, context=default, switchtype=national, and signalling=fxs_ks.</para>
<para>
We now having a working example of a zapata.conf with an FXS channel (1) and an FXO channel (4) that we can use. Channel 4 can be connected to an analog circuit such as might be provided by your phone company. You can plug an analog telephone directly into Channel 1.</para>
--- 148,152 ----
</informalexample>
<para>
! In the first example we listed channel 1 as an FXS channel. To create an FXO channel on the same TDM400P card, we list all the settings for the channel and then define the channel number. Instead of only having signalling be fxo_ks though we want the signalling to be fxs_ks. Because the other settings haven't been changed (signalling=fxs_ks replaces the previous value of signalling), they stay the same. Which means that while channel 1 takes on the values language=en, context=default, switchtype=national, and signalling=fxo_ks; channel 4 takes on the values language=en, context=default, switchtype=national, and signalling=fxs_ks.</para>
<para>
We now having a working example of a zapata.conf with an FXS channel (1) and an FXO channel (4) that we can use. Channel 4 can be connected to an analog circuit such as might be provided by your phone company. You can plug an analog telephone directly into Channel 1.</para>

cvs at sohoskyway.net (ji · Guest

Comments:
Update of /cvsroot/asterisk/docs/volume-one
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv5502/volume-one

Modified Files:
vm1chp4-channelconfig.xml
Log Message:
added a wee little thing to the SIP section
Index: vm1chp4-channelconfig.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/volume-one/vm1chp4-channelconfig.xml,v
retrieving revision 1.14
retrieving revision 1.15
diff -C2 -d -r1.14 -r1.15
*** vm1chp4-channelconfig.xml 1 Feb 2005 21:49:40 -0000 1.14
--- vm1chp4-channelconfig.xml 1 Feb 2005 22:57:34 -0000 1.15
***************
*** 201,208 ****
<programlisting>

! [general]
! port=5036 ; What port to use
! bindaddr=10.78.1.90 ; What IP address to bind to
! allow=all ; Allow the use of all audio codecs
register => username:secret@iaxtel.com

--- 201,208 ----
<programlisting>

! [general]
! port=5036 ; What port to use
! bindaddr=10.78.1.90 ; What IP address to bind to
! allow=all ; Allow the use of all audio codecs
register => username:secret@iaxtel.com

***************
*** 238,242 ****
SIP</title>
<para>
! The Session Initiation Protocol (SIP) is rapidly becoming the most widely supported VoIP protocol. Like IAX, SIP is pretty easy to set up. There are some gotcha's with the protocol though. Be aware that while your channels may be set up correctly, SIP does not handle NAT very well, and this can be a source of significant headaches.</para>
<sect2>
<title>
--- 238,242 ----
SIP</title>
<para>
! The Session Initiation Protocol (SIP) is rapidly becoming the most widely supported VoIP protocol. Like IAX, SIP is pretty easy to set up. There are some gotcha's with the protocol though. Be aware that while your channels may be set up correctly, SIP does not handle NAT very well, and this can be a source of significant headaches. To configure SIP, you will need to create a sip.conf file</para>
<sect2>
<title>