Differences between version 21 and predecessor to the previous major change of AsteriskSampleSetup.
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| Newer page: | version 21 | Last edited on Friday, April 22, 2005 3:39:04 pm | by SpinOff | Revert |
| Older page: | version 17 | Last edited on Thursday, February 19, 2004 9:11:34 pm | by KyleCarter | Revert |
@@ -19,9 +19,9 @@
PHONES2=SIP/2202
PHONES2VM=2202
PHONES3=SIP/2203
- PHONESVM
=2203
+ PHONES3VM
=2203
..etc..
;This will create a macro we will use in the dialling plan
@@ -30,8 +30,11 @@
exten => s,2,Playback(groovy)
exten => s,3,Playback(goodbye)
exten => s,4,Hangup
+; ----------------------------------------------%%%
+; DEFINE EXTENSIONS%%%
+; ----------------------------------------------%%%
[[home]
; Next, add an extension for voicemail .
; now if we dial 8, we can check voicemail.
;
@@ -56,11 +59,21 @@
;
exten => 2203,1,Dial(${PHONES3},20,Ttm)
exten => 2203,2,Macro(vmessage,${PHONES3VM})
exten => 2203,3,Hangup
+; ----------------------------------------------%%%
+; END DEFINE EXTENSIONS%%%
+; ----------------------------------------------%%%
-In sip.conf we want to setup the secrets so each phone can authenticate onto the line we just assigned in extensions.conf
+In sip.conf we want to setup the secrets so each phone can authenticate onto the line we just assigned in extensions.conf%%%
+WS - I find that encapsulating the SIP references with what extension it belongs to useful as well.%%%
+This makes reading the config file easier for someone who may need to alter it other than yourself%%%
+The same has been added to other parts of this example code.%%%
+;------------------------------------------------%%%
+; RECEPTIONIST - JULIE - EXT. 2202%%%
+; <extension use> - <users name> - <extension number>%%%
+;------------------------------------------------%%%
[[general]
;make sure to add this line, otherwise things dont work as proved by weeks of debugging
port=5060
;We want to configure each of the extensions we setup in extensions.conf like this one below (as a simple base)
@@ -74,9 +87,12 @@
callerid="Name you want displayed" <2202>
mailbox=2202
dtmfmode=rfc2833
nat=0
- ;context=home (this means that this unit is setup in the dial plan, which we have yet to setup)
+ ;context=home (this means that this unit is setup in the dial plan, which we have yet to setup)%%%
+;------------------------------------------------%%%
+; END RECEPTIONIST%%%
+;------------------------------------------------%%%
keep adding these sip.conf sections for each SIP unit we are adding to the network.
voicemail is the next section to setup, and just as in the other sections it doesnt take long
@@ -142,9 +158,9 @@
----
!Adding FWD to the callplan and reveiving incoming FWD calls.
-To add dialling from and to FreeWorldDialup
we need to make asterisk register to teh FWD servers with the correct password and username, we also need to make FWD ring the correct phones when asterisk receives a call from FWD. To so this we need to add the following lines to extensions.conf. This configuration assumes that you also have an extension with the same number as your FWD user ID if you do not already have this extension you should add it using the templates in the previous section.
+To add dialling from and to FreeWorldDialUp
we need to make asterisk register to teh FWD servers with the correct password and username, we also need to make FWD ring the correct phones when asterisk receives a call from FWD. To so this we need to add the following lines to extensions.conf. This configuration assumes that you also have an extension with the same number as your FWD user ID if you do not already have this extension you should add it using the templates in the previous section.
<in globals section>
FWDUSERID=<FWDUSERNUMBER>
MYNAME=<NAME>
@@ -205,8 +221,10 @@
host=fwd.pulver.com
dtmfmode=inband
context=home
nat=yes
- reinvite=no
canreinvite=no
disallow=all
allow=all
+
+
+AsteriskSipPhoneSetup
