Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv32711/configs

Modified Files:
        sip.conf.sample 
Log Message:
update sample configuration to reflect new options (issue #5357)


Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.71
retrieving revision 1.72
diff -u -d -r1.71 -r1.72
--- sip.conf.sample     27 Sep 2005 01:54:17 -0000      1.71
+++ sip.conf.sample     4 Oct 2005 19:05:40 -0000       1.72
@@ -40,6 +40,19 @@
                                ; ability to place SIP calls based on domain 
                                ; names to some other SIP users on the Internet
                                
+;domain=mydomain.tld           ; Set default domain for this host
+                               ; If configured, Asterisk will only allow
+                               ; INVITE and REFER to non-local domains
+                               ; Use "sip show domains" to list local domains
+;domain=mydomain.tld,mydomain-incoming
+                               ; Add domain and configure incoming context
+                               ; for external calls to this domain
+;domain=1.2.3.4                        ; Add IP address as local domain
+                               ; You can have several "domain" settings
+;allowexternalinvites=no       ; Disable INVITE and REFER to non-local domains
+                               ; Default is yes
+;autodomain=yes                        ; Turn this on to have Asterisk add 
local host
+                               ; name and local IP to domain list.
 ;pedantic=yes                  ; Enable slow, pedantic checking for Pingtel
                                ; and multiline formatted headers for strict
                                ; SIP compatibility (defaults to "no")
@@ -89,6 +102,10 @@
 ;compactheaders = yes          ; send compact sip headers.
 ;sipdebug = yes                        ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this 
configuration
+;subscribecontext = default    ; Set a specific context for SUBSCRIBE requests
+                               ; Useful to limit subscriptions to local 
extensions
+                               ; Settable per peer/user also
+;notifyringing = yes           ; Notify subscriptions on RINGING state
 
 ;
 ; If regcontext is specified, Asterisk will dynamically 
@@ -255,6 +272,7 @@
 ; amaflags                   amaflags
 ; call-limit                 call-limit
 ; restrictcid                restrictcid
+; subscribecontext           subscribecontext
 ;                             mailbox
 ;                             username
 ;                             template
@@ -338,17 +356,18 @@
 ;allow=gsm                     ; GSM consumes far less bandwidth than ulaw
 ;allow=ulaw
 ;allow=alaw
+;[EMAIL PROTECTED],[EMAIL PROTECTED]   ; Subscribe to status of multiple 
mailboxes
 
 
 ;[snom]
 ;type=friend                   ; Friends place calls and receive calls
 ;context=from-sip              ; Context for incoming calls from this user
 ;secret=blah
+;subscribecontext=localextensions      ; Only allow SUBSCRIBE for local 
extensions
 ;language=de                   ; Use German prompts for this user 
 ;host=dynamic                  ; This peer register with us
 ;dtmfmode=inband               ; Choices are inband, rfc2833, or info
 ;defaultip=192.168.0.59                ; IP used until peer registers
-;username=snom                 ; Username to use in INVITE until peer registers
 ;[EMAIL PROTECTED],2345      ; Mailbox(-es) for message waiting indicator
 ;vmexten=voicemail      ; dialplan extension to reach mailbox 
                         ; sets the Message-Account in the MWI notify message
@@ -365,6 +384,7 @@
 ;host=dynamic                  ; This peer register with us
 ;dtmfmode=rfc2833              ; Choices are inband, rfc2833, or info
 ;username=polly                        ; Username to use in INVITE until peer 
registers
+                               ; Normally you do NOT need to set this parameter
 ;disallow=all
 ;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
 ;progressinband=no             ; Polycom phones don't work properly with 
"never"
@@ -372,7 +392,6 @@
 
 ;[pingtel]
 ;type=friend
-;username=pingtel
 ;secret=blah
 ;host=dynamic
 ;insecure=port                 ; Allow matching of peer by IP address without 
matching port number
@@ -387,7 +406,6 @@
 
 ;[cisco1]
 ;type=friend
-;username=cisco1
 ;secret=blah
 ;qualify=200                   ; Qualify peer is no more than 200ms away
 ;nat=yes                       ; This phone may be natted
@@ -401,4 +419,6 @@
                                ; behind a NAT).
 ;defaultip=192.168.0.4         ; IP address to use until registration
 ;username=goran                        ; Username to use when calling this 
device before registration
+                               ; Normally you do NOT need to set this parameter
+;setvar=CUSTID=5678            ; Channel variable to be set for all calls from 
this device
 

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