Differences between version 2 and predecessor to the previous major change of AsteriskSipPhoneSetup.
Other diffs: Previous Revision, Previous Author, or view the Annotated Edit History
| Newer page: | version 2 | Last edited on Thursday, March 4, 2004 10:10:45 pm | by CraigBox | Revert |
| Older page: | version 1 | Last edited on Thursday, March 4, 2004 4:21:29 pm | by CraigMckenna | Revert |
@@ -1,69 +1,72 @@
-An Example setup for connecting from Asterisk to sipphone.com
+An Example setup for connecting from Asterisk to sipphone.com.
+!!extensions.conf
-__In extensions.conf
-[[Globals]%%%
-.%%%
-.%%%
-SIPPHONEUSERID=17473863406%%%
-%%%
-[[dialout]%%%
-.%%%
-.%%%
-include => sip-forced%%%
-include => from-sipphone%%%
-%%%
-%%%
-; Check to see if the called number starts with a "6" and%%%
-; if so, set the call parameters and bounce the call to the%%%
-; SipPhone.com SIP server.%%%
-;%%%
-; NOTE: Calls to unknown users will result in "invalid extension"%%%
-; message being played.%%%
-;%%%
-[[sip-forced]%%%
-%%%
-exten => _6.,1,SetCallerID(${SIPPHONEUSERID})%%%
-exten => _6.,2,SetCIDName(${MYNAME})%%%
-exten => _6.,3,Dial(SIP/${EXTEN:1}@sipphone)%%%
-exten => _6.,4,Playback(invalid)%%%
-exten => _6.,5,Hangup%%%
-%%%
-%%%
-; To receive calls inbound from SipPhone.com, we set the extension%%%
-; to our SipPhone user id, in this case from the SIPPHONEUSERID variable%%%
-; Changing the "Dial"%%%
-; directive to something like this:%%%
-; Dial(${PHONES1}&${PHONES2},15,Ttm)%%%
-; would cause both lines to ring%%%
-;%%%
-%%%
-[[from-sipphone]%%%
-exten => ${SIPPHONEUSERID},1,Dial(${PHONES1},30,Ttm)%%%
-exten => ${SIPPHONEUSERID},2,Voicemail2(u${PHONES1VM})%%%
-exten => ${SIPPHONEUSERID},3,Hangup%%%
-%%%
-%%%
-__In sip.conf%%%
-%%%
-[[general]%%%
-.%%%
-.%%%
-.%%%
-register=<YourSIPPhoneID>:<YourSIPPhonePassword>@proxy01.sipphone.com%%%
-%%%
+<verbatim>
+[[Globals]
+.
+.
+SIPPHONEUSERID=17473863406
+[[dialout]
+.
+.
+include => sip-forced
+include => from-sipphone
-[[sipphone]%%%
-type=friend%%%
-secret=<YourSIPPhonePassword>%%%
-username=<YourSIPPhoneID>%%%
-host=proxy01.sipphone.com%%%
-dtmfmode=inband%%%
-context=home%%%
-nat=yes%%%
-reinvite=no%%%
-canreinvite=no%%%
-disallow=all%%%
-allow=all%%%
+; Check to see if the called number starts with a "6" and
+; if so, set the call parameters and bounce the call to the
+; SipPhone.com SIP server.
+;
+; NOTE: Calls to unknown users will result in "invalid extension"
+; message being played.
+;
+[[sip-forced]
+
+exten => _6.,1,SetCallerID(${SIPPHONEUSERID})
+exten => _6.,2,SetCIDName(${MYNAME})
+exten => _6.,3,Dial(SIP/${EXTEN:1}@sipphone)
+exten => _6.,4,Playback(invalid)
+exten => _6.,5,Hangup
+
+
+; To receive calls inbound from SipPhone.com, we set the extension
+; to our SipPhone user id, in this case from the SIPPHONEUSERID variable
+; Changing the "Dial"
+; directive to something like this:
+; Dial(${PHONES1}&${PHONES2},15,Ttm)
+; would cause both lines to ring
+;
+
+[[from-sipphone]
+exten => ${SIPPHONEUSERID},1,Dial(${PHONES1},30,Ttm)
+exten => ${SIPPHONEUSERID},2,Voicemail2(u${PHONES1VM})
+exten => ${SIPPHONEUSERID},3,Hangup
+</verbatim>
+
+!!sip.conf
+
+<verbatim>
+[[general]
+.
+.
+.
+register=<YourSIPPhoneID>:<YourSIPPhonePassword>@proxy01.sipphone.com
+
+
+
+
+
[[sipphone]
+type=friend
+secret=<YourSIPPhonePassword>
+username=<YourSIPPhoneID>
+host=proxy01.sipphone.com
+dtmfmode=inband
+context=home
+nat=yes
+reinvite=no
+canreinvite=no
+disallow=all
+allow=all
+</verbatim>
