git-svn-id: https://svn.code.sf.net/p/freetel/code@858 01035d8c-6547-0410-b346-abe4f9...

diff --git a/mini-asterisk-gui2/todo 2012_10.txt b/mini-asterisk-gui2/todo 2012_10.txt
index 886fc1e7456aba60f4d6091aa241fa35d227ea30..19efefacc8d1fd88146ba9901b632b7d78d9fbf7 100644 (file)
--- a/mini-asterisk-gui2/todo 2012_10.txt
+++ b/mini-asterisk-gui2/todo 2012_10.txt
@@ -69,5 +69,6 @@ Oct 29 2012
 IP provider page work\r
        sip.conf modified with new keywords\r
        install script changed to update sip.conf\r
+       Are sip provider entries ok to leav uncommented?...they have dummy hosts\r
        \r
 Notice ip phone (ekiga from pc) registrations stay on ip04 after phone has disappeared
\ No newline at end of file

diff --git a/mini-asterisk-gui2/voiplines.js b/mini-asterisk-gui2/voiplines.js
index b0dd9fdb264037918a8eedf69e81f5ff88124972..b8c58d74cd294bba55eba313a6e6e81abc550613 100644 (file)
--- a/mini-asterisk-gui2/voiplines.js
+++ b/mini-asterisk-gui2/voiplines.js
@@ -69,13 +69,31 @@ function onClickApply() {
        // You will need to cascade calls to downloadUrl
 
        // take provider, username, password and host. 
-
-       // 1. Save old sip.conf
-       // 2. Make new sip.conf with provider uncommented
-       //  2.1 insert register command
-       //      2.2     uncomment sip trunk entry
-       //  2.3 update sip trunk details
+       var user = "";
+       var host = "";
+       var passwd = "";
+       
+       // 1. Save sip.conf     
+       
+       
+       user = document.getElementById('user').value;
+       host = document.getElementById('host').value;
+       passwd = document.getElementById('pass').value
+       
+       
+       // 2. Update sip.conf
+       //  2.1 insert register command  
+       //          ;register => 1234:password@mysipprovider.com   becomes     register => trev:password@192.168.1.30 
+       
+       
+       
+       //  2.2 update sip trunk details  ... user pw host
+       //          [usersip]             becomes       [trev]
+       //          username=usersip                     username=trev
+       //          secret=passwordsip                   secret=password
+       //          host=hostsip                         host=192.168.1.30
        // 3. Modify extensions.conf for new provider
+       //          exten => _1.,1,Dial(SIP/voip/${EXTEN:1})    becomes     exten => _1.,1,Dial(SIP/trev/${EXTEN:1}) 
        // 4. asterisk sip reload ....ORDER? beardy has this at position 4
        // 5. asterisk dialplan reload