Differences between version 6 and predecessor to the previous major change of AsteriskSipPhoneSetup.
Other diffs: Previous Revision, Previous Author, or view the Annotated Edit History
| Newer page: | version 6 | Last edited on Tuesday, November 23, 2004 4:54:19 pm | by JohnMcPherson | Revert |
| Older page: | version 1 | Last edited on Thursday, March 4, 2004 4:21:29 pm | by CraigMckenna | Revert |
@@ -1,69 +1,82 @@
-An Example setup for connecting from Asterisk to sipphone.com
+An Example setup for connecting from Asterisk to sipphone.com.
+Note: If you are behind a router/firewall, you'll want to forward the ports 5004, 5060, and 69 to your local ip with the asterisk server on it.
-__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 => ${SIPPHONEUSERID},3,Hangup%%%
-%%%
-%%%
-__In sip.conf%%%
-%%%
-[[general]%%%
-.%%%
-.%%%
-.%%%
-register=<YourSIPPhoneID>:<YourSIPPhonePassword>@proxy01.sipphone.com%%%
-%%%
+!!
extensions.conf
+<verbatim>
+[globals]
+.
+.
+SIPPHONEUSERID=1747xxxXXXX
+MYNAME=Bob
+[[dialout]
+.
+.
+include => sip-forced
+include => from-sipphone
-[[sipphone]%%%
-type=friend%%%
-secret=<YourSIPPhonePassword>%%%
-username=<YourSIPPhoneID>%%%
-host=proxy01.sipphone.com%%%
-dtmfmode=inband%%%
-context=home%%%
-nat=yes%%%
-reinvite=no%%%
-canreinvite=no%%%
-disallow=all%%%
-allow=all%%%
+; Check to see if the called number starts with a "6" and
+; if so, set the call parameters and bounce the call to the
+; SipPhone.com SIP server.
+;
+; NOTE: Calls to unknown users will result in "invalid extension"
+; message being played.
+;
+
[[sip-forced]
+
+exten => _6.,1,SetCallerID(${SIPPHONEUSERID})
+exten => _6.,2,SetCIDName(${MYNAME})
+exten => _6.,3,Dial(SIP/${EXTEN:1}@proxy01.
sipphone.com)
+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/<extension>
+.
+.
+;be sure to set your external ip if you're behind a router.
+;Especially if you want to use sipphone minutes
+externip=xxx.xxx.xxx.xxx
+.
+.
+;uncomment this line as it should be in your default sip.conf
+localnet=192.168../255.255..; All RFC 1918 addresses are local networks
+
+
+
+;Note: the name is [proxy01.sipphone.com] so that incoming calls work correctly
+[proxy01.sipphone.com
]
+type=friend
+secret=<YourSIPPhonePassword>
+username=<YourSIPPhoneID>
+host=proxy01.sipphone.com
+dtmfmode=inband
+context=dialout ;or your context which includes the dialing rules
+nat=yes
+qualify=no
+reinvite=no
+canreinvite=no
+</verbatim>
