Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b

21 files changed, 60 insertions, 59 deletions

diff --git a/doc/asterisk.8 b/doc/asterisk.8
index 876721a93..76daafc67 100644
--- a/doc/asterisk.8
+++ b/doc/asterisk.8
@@ -122,10 +122,10 @@ Allows to specify the socket file to be used to connect to the
 Asterisk console. Used in conjunction with \fB-r\fR or \fB-R\fR.
 .TP
 \fB-I\fR
-Enable internal timing if Zaptel timer is available
+Enable internal timing if DAHDI timer is available
 The default behaviour is that outbound packets are phase locked
 to inbound packets. Enabling this switch causes them to be
-locked to the internal Zaptel timer instead.
+locked to the internal DAHDI timer instead.
 .TP
 \fB-t\fR
 When recording files, write them first into a temporary holding directory, 
diff --git a/doc/asterisk.sgml b/doc/asterisk.sgml
index ebcecfc06..6a7823f71 100644
--- a/doc/asterisk.sgml
+++ b/doc/asterisk.sgml
@@ -157,10 +157,10 @@
 		<term>-I</term>
 		<listitem>
 			<para>
-			Enable internal timing if Zaptel timing is available.
+			Enable internal timing if DAHDI timing is available.
 			The default behaviour is that outbound packets are phase locked
 			to inbound packets. Enabling this switch causes them to be
-			locked to the internal Zaptel timer instead.
+			locked to the internal DAHDI timer instead.
 			</para>
 		</listitem>
 	</varlistentry>
diff --git a/doc/backtrace.txt b/doc/backtrace.txt
index ce39b0a3e..caf1579ce 100644
--- a/doc/backtrace.txt
+++ b/doc/backtrace.txt
@@ -97,7 +97,7 @@ You would see output similar to:
 #10 0x000000a0 in ?? ()
 #11 0x000000a0 in ?? ()
 #12 0x00000000 in ?? ()
-#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262
+#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
 #14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
 #15 0xbcdffbe0 in ?? ()
 #16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
@@ -135,7 +135,7 @@ No symbol table info available.
 No symbol table info available.
 #12 0x00000000 in ?? ()
 No symbol table info available.
-#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262
+#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
         f = (struct ast_frame *) 0x8180bf8
         trans = (struct ast_trans_pvt *) 0x0
 #14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
@@ -173,7 +173,7 @@ Thread 1 (process 26252):
 #10 0x000000a0 in ?? ()
 #11 0x000000a0 in ?? ()
 #12 0x00000000 in ?? ()
-#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262
+#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
 #14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
 #15 0xbcdffbe0 in ?? ()
 #16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
diff --git a/doc/janitor-projects.txt b/doc/janitor-projects.txt
index 65416cc52..a43f9c957 100644
--- a/doc/janitor-projects.txt
+++ b/doc/janitor-projects.txt
@@ -17,7 +17,7 @@
  -- Convert all existing uses of astobj.h to astobj2.h
     -- (chan_sip already in progress in a branch)
 
- -- There are many places where large character buffers are allocated in structures.  There is a new system for string handling that uses dynamically allocatted memory pools which is documented in include/asterisk/stringfields.h.  Examples of where they are currently used are the ast_channel structure defined in include/asterisk/channel.h, some structures in chan_sip.c, and chan_zap.c.
+ -- There are many places where large character buffers are allocated in structures.  There is a new system for string handling that uses dynamically allocatted memory pools which is documented in include/asterisk/stringfields.h.  Examples of where they are currently used are the ast_channel structure defined in include/asterisk/channel.h, some structures in chan_sip.c, and chan_dahdi.c.
 
  -- There is a convenient set of macros defined in include/asterisk/linkedlists.h for handling linked lists.  However, there are some open-coded lists throughout the code.  Converting linked lists to use these macros will make list handling more consistent and reduce the possibility of coding errors.
 
diff --git a/doc/manager_1_1.txt b/doc/manager_1_1.txt
index 3d0440f0a..6feede773 100644
--- a/doc/manager_1_1.txt
+++ b/doc/manager_1_1.txt
@@ -99,14 +99,14 @@ Changes to manager version 1.1:
 
 - The MeetmeJoin now has caller ID name and Caller ID number fields (like MeetMeLeave)
 
-- Action ZapShowChannels
+- Action DAHDIShowChannels
 	Header changes
-	- Channel:	-> ZapChannel
+	- Channel:	-> DAHDIChannel
 	For active channels, the Channel: and Uniqueid: headers are added
-	You can now add a "ZapChannel: " argument to zapshowchannels actions
+	You can now add a "DAHDIChannel: " argument to DAHDIshowchannels actions
 	to only get information about one channel.
 
-- Event ZapShowChannelsComplete
+- Event DAHDIShowChannelsComplete
 	New header
 	- (new)		-> Items: 	Reports number of channels reported
 
diff --git a/doc/osp.txt b/doc/osp.txt
index 763bb3871..6bad3a25c 100644
--- a/doc/osp.txt
+++ b/doc/osp.txt
@@ -25,7 +25,7 @@ Revision History	                        3
 3.1.1	Build Asterisk with OSP Toolkit	8
 3.1.2	osp.conf	                        8
 3.1.3	extensions.conf	                 10
-3.1.4	zapata/sip/iax/h323/ooh323.conf    13
+3.1.4	dahdi/sip/iax/h323/ooh323.conf    13
 3.2	OSP Dial Plan Functions	           13
 3.2.1	OSPAuth	                       13
 3.2.2	OSPLookup	                       14
@@ -447,7 +447,7 @@ exten => s,300,GoTo(1000)
 ; --------------------------------------------------------------
 exten => s,1000,MacroExit
 
-3.1.4 zapata/sip/iax/h323/ooh323.conf
+3.1.4 dahdi/sip/iax/h323/ooh323.conf
 There is no configuration required for OSP.
 
 3.2 OSP Dial Plan Functions
diff --git a/doc/sms.txt b/doc/sms.txt
index fe0ec8d85..d23f55aac 100644
--- a/doc/sms.txt
+++ b/doc/sms.txt
@@ -5,7 +5,7 @@ message centres using ETSI ES 201 912 protocol 1 FSK messaging over analog calls
 
 Basically it allows sending and receiving of text messages over the PSTN. It is 
 compatible with BT Text service in the UK and works on ISDN and PSTN lines. It is 
-designed to connect to an ISDN or zap interface directly and uses FSK so would 
+designed to connect to an ISDN or DAHDI interface directly and uses FSK so would 
 probably not work over any sort of compressed link (like a VoIP call using GSM codec).
 
 Typical applications include:-
diff --git a/doc/ss7.txt b/doc/ss7.txt
index 632035df0..1751050ac 100644
--- a/doc/ss7.txt
+++ b/doc/ss7.txt
@@ -31,6 +31,7 @@ is how to check them out from the public subversion server.
 
 These are the commands you would type to install them:
 
+#jpeeler: REVISIT
 `svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4`
 `cd zaptel-1.4`
 `make; make install`
diff --git a/doc/tex/ael.tex b/doc/tex/ael.tex
index b76088fa0..c483947d8 100644
--- a/doc/tex/ael.tex
+++ b/doc/tex/ael.tex
@@ -44,10 +44,10 @@ as well as the Expression syntax, and the Variable syntax.
 
 \section{Asterisk in a Nutshell}
 
-Asterisk acts as a server. Devices involved in telephony, like Zapata
+Asterisk acts as a server. Devices involved in telephony, like DAHDI
 cards, or Voip phones, all indicate some context that should be
 activated in their behalf. See the config file formats for IAX, SIP,
-zapata.conf, etc. They all help describe a device, and they all
+dahdi.conf, etc. They all help describe a device, and they all
 specify a context to activate when somebody picks up a phone, or a
 call comes in from the phone company, or a voip phone, etc.
 
@@ -707,7 +707,7 @@ Global variables are set in their own block.
 \begin{verbatim}
 globals {
     CONSOLE=Console/dsp;
-    TRUNK=Zap/g2;
+    TRUNK=DAHDI/g2;
 }
 \end{verbatim}
 \end{astlisting}
diff --git a/doc/tex/app-sms.tex b/doc/tex/app-sms.tex
index aa515f61a..ec65a66f9 100644
--- a/doc/tex/app-sms.tex
+++ b/doc/tex/app-sms.tex
@@ -151,7 +151,7 @@ exten = _X./_80058752[0-8]0,1,Goto(smsmtrx,${EXTEN}-${CALLERID(num):8:1},1)
 \end{astlisting}
 
    Alternatively, if you have the correct national prefix on incoming
-   CLI, e.g. using zaphfc, you might use:
+   CLI, e.g. using dahdi_hfc, you might use:
 \begin{astlisting}
 \begin{verbatim}
 exten = _X./08005875290,1,Goto(smsmtrx,${EXTEN},1)
diff --git a/doc/tex/backtrace.tex b/doc/tex/backtrace.tex
index f43f42327..fd1c39d00 100644
--- a/doc/tex/backtrace.tex
+++ b/doc/tex/backtrace.tex
@@ -111,7 +111,7 @@ You would see output similar to:
 #10 0x000000a0 in ?? ()
 #11 0x000000a0 in ?? ()
 #12 0x00000000 in ?? ()
-#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262
+#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
 #14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
 #15 0xbcdffbe0 in ?? ()
 #16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
@@ -152,7 +152,7 @@ No symbol table info available.
 No symbol table info available.
 #12 0x00000000 in ?? ()
 No symbol table info available.
-#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262
+#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
         f = (struct ast_frame *) 0x8180bf8
         trans = (struct ast_trans_pvt *) 0x0
 #14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
@@ -194,7 +194,7 @@ Thread 1 (process 26252):
 #10 0x000000a0 in ?? ()
 #11 0x000000a0 in ?? ()
 #12 0x00000000 in ?? ()
-#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262
+#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
 #14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
 #15 0xbcdffbe0 in ?? ()
 #16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
diff --git a/doc/tex/channelvariables.tex b/doc/tex/channelvariables.tex
index ae5b1974b..ae28818d3 100644
--- a/doc/tex/channelvariables.tex
+++ b/doc/tex/channelvariables.tex
@@ -871,7 +871,7 @@ ${MEETME_RECORDINGFILE}      Name of file for recording a conference with
                              the "r" option
 ${MEETME_RECORDINGFORMAT}    Format of file to be recorded
 ${MEETME_EXIT_CONTEXT}       Context for exit out of meetme meeting
-${MEETME_AGI_BACKGROUND}     AGI script for Meetme (zap only)
+${MEETME_AGI_BACKGROUND}     AGI script for Meetme (DAHDI only)
 ${MEETMESECS}              * Number of seconds a user participated in a MeetMe conference
 ${CONF_LIMIT_TIMEOUT_FILE}   File to play when time is up.  Used with the L() option.
 ${CONF_LIMIT_WARNING_FILE}   File to play as warning if 'y' is defined.
@@ -903,7 +903,7 @@ ${DUNDTECH}       * The Technology of the result from a call to DUNDiLookup()
 ${DUNDDEST}       * The Destination of the result from a call to DUNDiLookup()
 \end{verbatim}
 
-\subsection{chan\_zap}
+\subsection{chan\_dahdi}
 \begin{verbatim}
 ${ANI2}               * The ANI2 Code provided by the network on the incoming call.
                         (ie, Code 29 identifies call as a Prison/Inmate Call)
diff --git a/doc/tex/configuration.tex b/doc/tex/configuration.tex
index 9257a86ba..712b6b308 100644
--- a/doc/tex/configuration.tex
+++ b/doc/tex/configuration.tex
@@ -20,7 +20,7 @@ Asterisk configuration files are defined as follows:
 \end{verbatim}
 \end{astlisting}
 
-In some files, (e.g. mgcp.conf, zapata.conf and agents.conf), the syntax
+In some files, (e.g. mgcp.conf, dahdi.conf and agents.conf), the syntax
 is a bit different. In these files the syntax is as follows:
 	
 \begin{astlisting}
diff --git a/doc/tex/enum.tex b/doc/tex/enum.tex
index 9a3384d46..9341a6ea1 100644
--- a/doc/tex/enum.tex
+++ b/doc/tex/enum.tex
@@ -286,7 +286,7 @@ ENUMLOOKUP function calls.
 ;
 exten => _011.,1,Set(enumresult=${ENUMLOOKUP(+${EXTEN:3})})
 exten => _011.,n,Dial(SIP/${enumresult})
-exten => _011.,n,Dial(Zap/g1/${EXTEN})
+exten => _011.,n,Dial(DAHDI/g1/${EXTEN})
 ;
 ; end example 1
 
@@ -302,7 +302,7 @@ exten => _011.,n,While($["${counter}"<"${sipcount}"])
 exten => _011.,n,Set(counter=$[${counter}+1])
 exten => _011.,n,Dial(SIP/${ENUMLOOKUP(+${EXTEN:3},sip,,${counter})})
 exten => _011.,n,EndWhile
-exten => _011.,n,Dial(Zap/g1/${EXTEN})
+exten => _011.,n,Dial(DAHDI/g1/${EXTEN})
 ;
 ; end example 2
 
@@ -312,7 +312,7 @@ exten => _011.,n,Dial(Zap/g1/${EXTEN})
 ;  14102241145 or 437203001721)
 ; Search through e164.arpa and then also search through e164.org
 ;  to see if there are any valid SIP or IAX termination capabilities.
-;  If none, send call out via Zap channel 1.
+;  If none, send call out via DAHDI channel 1.
 ;
 ; Start first with e164.arpa zone...
 ;
@@ -346,8 +346,8 @@ exten => _X.,21,GotoIf($["${counter}"<"${iaxcount}"]?19:22)
 ;
 ; ...then send out PRI.
 ;
-exten => _X.,22,NoOp("No valid entries in e164.org for ${EXTEN} - sending out via Zap")
-exten => _X.,23,Dial(Zap/g1/${EXTEN})
+exten => _X.,22,NoOp("No valid entries in e164.org for ${EXTEN} - sending out via DAHDI")
+exten => _X.,23,Dial(DAHDI/g1/${EXTEN})
 ;
 ; end example 3
 
diff --git a/doc/tex/hardware.tex b/doc/tex/hardware.tex
index 30fa587aa..a678167de 100644
--- a/doc/tex/hardware.tex
+++ b/doc/tex/hardware.tex
@@ -4,12 +4,12 @@ A PBX is only really useful if you can get calls into it.  Of course, you
 can use Asterisk with VoIP calls (SIP, H.323, IAX, etc.), but you can also 
 talk to the real PSTN through various cards.
 
-Supported Hardware is divided into two general groups:  Zaptel devices and 
-non-zaptel devices.  The Zaptel compatible hardware supports pseudo-TDM 
-conferencing and all call features through chan\_zap, whereas non-zaptel 
+Supported Hardware is divided into two general groups:  DAHDI devices and 
+non-DAHDI devices.  The DAHDI compatible hardware supports pseudo-TDM 
+conferencing and all call features through chan\_dahdi, whereas non-DAHDI 
 compatible hardware may have different features.
 
-\subsection{Zaptel compatible hardware}
+\subsection{DAHDI compatible hardware}
 
 \begin{itemize}
 \item Digium, Inc. (Primary Developer of Asterisk)
@@ -37,7 +37,7 @@ compatible hardware may have different features.
   \end{itemize}
 \end{itemize}
 
-\subsection{Non-zaptel compatible hardware}
+\subsection{Non-DAHDI compatible hardware}
 
 \begin{itemize}
   \item QuickNet, Inc. 
diff --git a/doc/tex/localchannel.tex b/doc/tex/localchannel.tex
index ab42606f7..528421e0b 100644
--- a/doc/tex/localchannel.tex
+++ b/doc/tex/localchannel.tex
@@ -61,10 +61,10 @@ exten => s,4,Hangup
 exten => 200,1,Dial(sip/blah)
 exten => 200,102,VoiceMail(${EXTEN}@default)
 
-exten => 201,1,Dial(zap/1)
+exten => 201,1,Dial(DAHDI/1)
 exten => 201,102,VoiceMail(${EXTEN}@default)
 
-exten => _0.,1,Dial(Zap/g1/${EXTEN:1}) ; outgoing calls with 0+number
+exten => _0.,1,Dial(DAHDI/g1/${EXTEN:1}) ; outgoing calls with 0+number
 \end{verbatim}
 \end{astlisting}
 
diff --git a/doc/tex/manager.tex b/doc/tex/manager.tex
index c3b567bd4..1f9fa1495 100644
--- a/doc/tex/manager.tex
+++ b/doc/tex/manager.tex
@@ -99,7 +99,7 @@ Redirect with ExtraChannel:
 
 \begin{verbatim}
   Action: Redirect
-  Channel: Zap/1-1
+  Channel: DAHDI/1-1
   ExtraChannel: SIP/3064-7e00 (varies)
   Exten: 680
   Priority: 1
@@ -147,7 +147,7 @@ the mailing list archives and the documentation page on the
  Channel: <dialstring>    -- Dialstring in Originate
  Channel: <tech/[peer/username]> -- Channel in Registry events (SIP, IAX2)
  Channel: <tech>          -- Technology (SIP/IAX2 etc) in Registry events
- ChannelType:             -- Tech: SIP, IAX2, ZAP, MGCP etc
+ ChannelType:             -- Tech: SIP, IAX2, DAHDI, MGCP etc
  Channel1:                -- Link channel 1
  Channel2:                -- Link channel 2
  ChanObjectType:          -- "peer", "user"
diff --git a/doc/tex/privacy.tex b/doc/tex/privacy.tex
index f8bf698f6..17cf6a12b 100644
--- a/doc/tex/privacy.tex
+++ b/doc/tex/privacy.tex
@@ -169,11 +169,11 @@ There are some variations, and these will be explained in due course.
 To use these options, set your Dial to something like:
 \begin{astlisting}
 \begin{verbatim}
-exten => 3,3,Dial(Zap/5r3&Zap/6r3,35,tmPA(beep))
+exten => 3,3,Dial(DAHDI/5r3&DAHDI/6r3,35,tmPA(beep))
     or
-exten => 3,3,Dial(Zap/5r3&Zap/6r3,35,tmP(something)A(beep))
+exten => 3,3,Dial(DAHDI/5r3&DAHDI/6r3,35,tmP(something)A(beep))
     or
-exten => 3,3,Dial(Zap/5r3&Zap/6r3,35,tmpA(beep))
+exten => 3,3,Dial(DAHDI/5r3&DAHDI/6r3,35,tmpA(beep))
 \end{verbatim}
 \end{astlisting}
 
diff --git a/doc/tex/queues-with-callback-members.tex b/doc/tex/queues-with-callback-members.tex
index 36e642845..30051eecf 100644
--- a/doc/tex/queues-with-callback-members.tex
+++ b/doc/tex/queues-with-callback-members.tex
@@ -466,7 +466,7 @@ context agents
 
 In the above, the variables \$\{RAQUEL\}, etc stand for
 actual devices to ring that person's
-phone (like Zap/37).
+phone (like DAHDI/37).
 
 The 8010, 8011, and 8013 extensions are purely for transferring
 incoming callers to queues. For instance, a customer service
diff --git a/doc/tex/security.tex b/doc/tex/security.tex
index 4eb4e1095..975f4bdde 100644
--- a/doc/tex/security.tex
+++ b/doc/tex/security.tex
@@ -57,15 +57,15 @@ in the appropriate section.  A well designed PBX might look like this:
 \begin{astlisting}
 \begin{verbatim}
 [longdistance]
-exten => _91NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
+exten => _91NXXNXXXXXX,1,Dial(DAHDI/g2/${EXTEN:1})
 include => local
 
 [local]
-exten => _9NXXNXXX,1,Dial(Zap/g2/${EXTEN:1})
+exten => _9NXXNXXX,1,Dial(DAHDI/g2/${EXTEN:1})
 include => default
 
 [default]
-exten => 6123,Dial(Zap/1)
+exten => 6123,Dial(DAHDI/1)
 \end{verbatim}
 \end{astlisting}
 
diff --git a/doc/tex/sla.tex b/doc/tex/sla.tex
index afafd2ae4..844a4f2a7 100644
--- a/doc/tex/sla.tex
+++ b/doc/tex/sla.tex
@@ -23,7 +23,7 @@ IP channels.
 An SLA system is built up of virtual trunks and stations mapped to real
 Asterisk devices.  The configuration for all of this is done in three
 different files: extensions.conf, sla.conf, and the channel specific
-configuration file such as sip.conf or zapata.conf.
+configuration file such as sip.conf or dahdi.conf.
 
 \subsection{Dialplan}
 
@@ -55,21 +55,21 @@ Please refer to the examples section for full dialplan samples for SLA.
 An SLA trunk is a mapping between a virtual trunk and a real Asterisk device.
 This device may be an analog FXO line, or something like a SIP trunk.  A trunk
 must be configured in two places.  First, configure the device itself in the
-channel specific configuration file such as zapata.conf or sip.conf.  Once the
+channel specific configuration file such as dahdi.conf or sip.conf.  Once the
 trunk is configured, then map it to an SLA trunk in sla.conf.
 \begin{astlisting}
 \begin{verbatim}
 [line1]
 type=trunk
-device=Zap/1
+device=DAHDI/1
 \end{verbatim}
 \end{astlisting}
 
 Be sure to configure the trunk's context to be the same one that is set for the
 "autocontext" option in sla.conf if automatic dialplan configuration is used.
-This would be done in the regular device entry in zapata.conf, sip.conf, etc.
+This would be done in the regular device entry in dahdi.conf, sip.conf, etc.
 Note that the automatic dialplan generation creates the SLATrunk() extension
-at extension 's'.  This is perfect for Zap channels that are FXO trunks, for
+at extension 's'.  This is perfect for DAHDI channels that are FXO trunks, for
 example.  However, it may not be good enough for an IP trunk, since the call
 coming in over the trunk may specify an actual number.
 
@@ -173,12 +173,12 @@ sla.conf:
 \begin{verbatim}
 [line1]
 type=trunk
-device=Zap/1
+device=DAHDI/1
 autocontext=line1
 
 [line2]
 type=trunk
-device=Zap/2
+device=DAHDI/2
 autocontext=line2
 
 [station](!)
@@ -199,8 +199,8 @@ device=SIP/station3
 \end{astlisting}
 
 With this configuration, the dialplan is generated automatically.  The first
-zap channel should have its context set to "line1" and the second should be
-set to "line2" in zapata.conf.  In sip.conf, station1, station2, and station3
+DAHDI channel should have its context set to "line1" and the second should be
+set to "line2" in dahdi.conf.  In sip.conf, station1, station2, and station3
 should all have their context set to "sla\_stations".
 
 For reference, here is the automatically generated dialplan for this situation:
@@ -241,10 +241,10 @@ phone system.  The voicemail box number used in this example is 1234, which
 would be configured in voicemail.conf.
 
 For this example, assume that there are 2 trunks and 3 stations.  The trunks
-are Zap/1 and Zap/2.  The stations are SIP/station1, SIP/station2, and
+are DAHDI/1 and DAHDI/2.  The stations are SIP/station1, SIP/station2, and
 SIP/station3.
 
-In zapata.conf, channel 1 has context=line1 and channel 2 has context=line2.
+In dahdi.conf, channel 1 has context=line1 and channel 2 has context=line2.
 
 In sip.conf, all three stations are configured with context=sla\_stations.
 
@@ -297,12 +297,12 @@ exten => s,2,Macro(slaline,line2)
 
 [line1_outbound]
 exten => disa,1,Disa(no-password,line1_outbound)
-exten => _1NXXNXXXXXX,1,Dial(Zap/1/${EXTEN})
+exten => _1NXXNXXXXXX,1,Dial(DAHDI/1/${EXTEN})
 exten => 8500,1,VoicemailMain(1234)
 
 [line2_outbound]
 exten => disa,1,Disa(no-password|line2_outbound)
-exten => _1NXXNXXXXXX,1,Dial(Zap/2/${EXTEN})
+exten => _1NXXNXXXXXX,1,Dial(DAHDI/2/${EXTEN})
 exten => 8500,1,VoicemailMain(1234)
 
 [sla_stations]