From 65c9df9b7f7d1e7e302f8f76253691a62a0fb34c Mon Sep 17 00:00:00 2001
From: drowe67 <drowe67@01035d8c-6547-0410-b346-abe4f91aad63>
Date: Tue, 5 Jan 2010 08:09:04 +0000
Subject: [PATCH] first pass at phone GUI

git-svn-id: https://svn.code.sf.net/p/freetel/code@95 01035d8c-6547-0410-b346-abe4f91aad63
---
 easy-asterisk-gui/Makefile        |   4 +-
 easy-asterisk-gui/dashboard.sh    |   4 +-
 easy-asterisk-gui/extensions.conf |  56 +--
 easy-asterisk-gui/ipphone.png     | Bin 0 -> 2177 bytes
 easy-asterisk-gui/menu.html       |   2 +-
 easy-asterisk-gui/network.sh      |   2 +-
 easy-asterisk-gui/phone.pl        | 121 +++++++
 easy-asterisk-gui/phone.png       | Bin 0 -> 2314 bytes
 easy-asterisk-gui/phone.sh        |  27 +-
 easy-asterisk-gui/phoneline.jpg   | Bin 0 -> 1198 bytes
 easy-asterisk-gui/sip.conf        | 546 ++++++++++++++++++++++++++++++
 easy-asterisk-gui/tooltips.html   |  13 +
 easy-asterisk-gui/voipline.jpg    | Bin 0 -> 1809 bytes
 13 files changed, 726 insertions(+), 49 deletions(-)
 create mode 100644 easy-asterisk-gui/ipphone.png
 create mode 100755 easy-asterisk-gui/phone.pl
 create mode 100644 easy-asterisk-gui/phone.png
 create mode 100644 easy-asterisk-gui/phoneline.jpg
 create mode 100644 easy-asterisk-gui/sip.conf
 create mode 100644 easy-asterisk-gui/voipline.jpg

diff --git a/easy-asterisk-gui/Makefile b/easy-asterisk-gui/Makefile
index 9c0e4bfd..196a21e8 100644
--- a/easy-asterisk-gui/Makefile
+++ b/easy-asterisk-gui/Makefile
@@ -9,5 +9,5 @@ TESTIP=192.168.1.150
 #	rcp index.html root@$(TESTIP):/var/lib/asterisk/static-http/
 
 test:
-	rcp tooltips.html phone.sh phone.js tick.png cross.png banner.html menu.html check_loggedin.sh logout.sh login.sh dashboard.sh dashboard.js network.sh network.js set_network.sh tooltip.js tooltip.css root@$(TESTIP):/www/cgi-bin
-	rcp extensions.conf root@$(TESTIP):/etc/asterisk
+	rcp voipline.jpg phone.png phoneline.jpg ipphone.png phone.pl tooltips.html phone.sh phone.js tick.png cross.png banner.html menu.html check_loggedin.sh logout.sh login.sh dashboard.sh dashboard.js network.sh network.js set_network.sh tooltip.js tooltip.css root@$(TESTIP):/www/cgi-bin
+	rcp sip.conf extensions.conf root@$(TESTIP):/etc/asterisk
diff --git a/easy-asterisk-gui/dashboard.sh b/easy-asterisk-gui/dashboard.sh
index 0b56f36e..81daa21c 100644
--- a/easy-asterisk-gui/dashboard.sh
+++ b/easy-asterisk-gui/dashboard.sh
@@ -34,7 +34,7 @@ cat << EOF
 EOF
 
 cat tooltips.html
-echo "<table align="center" width=800 border=0>"
+echo '<table align="center" width=800 border=0>'
 cat banner.html
 echo "    <tr>"
 cat menu.html    
@@ -44,7 +44,7 @@ cat <<EOF
 
     <form action="/cgi-bin/set_network.sh" onsubmit="return validate_form(this)" method="get">
     <table align="right" width=600 border=0>
-      <tr><td colspan="2" align="left" valign="top" ><h2>Dashboard</h2></td></tr>
+      <tr><td colspan="3" align="left" valign="top" ><h2>Dashboard</h2></td></tr>
       <tr onMouseOver="popUp(event,'network_internet')" onmouseout="popUp(event,'network_internet')">
 	  <td>Internet Connection:</td>
 	  <td><div id="internet" >
diff --git a/easy-asterisk-gui/extensions.conf b/easy-asterisk-gui/extensions.conf
index d1012ab9..48c59c1f 100644
--- a/easy-asterisk-gui/extensions.conf
+++ b/easy-asterisk-gui/extensions.conf
@@ -19,36 +19,36 @@ priorityjumping = no
 ; Pre-configured analog extensions, depends on IP0X model and what modules
 ; are installed.  Some of these may map to FXO ports
 
-exten => 6001,n,Dial(Zap/1)
-exten => 6002,n,Dial(Zap/2)
-exten => 6003,n,Dial(Zap/3)
-exten => 6004,n,Dial(Zap/4)
-exten => 6005,n,Dial(Zap/5)
-exten => 6006,n,Dial(Zap/6)
-exten => 6007,n,Dial(Zap/7)
-exten => 6008,n,Dial(Zap/8)
+exten => 6001,1,Dial(Zap/1)
+exten => 6002,1,Dial(Zap/2)
+exten => 6003,1,Dial(Zap/3)
+exten => 6004,1,Dial(Zap/4)
+exten => 6005,1,Dial(Zap/5)
+exten => 6006,1,Dial(Zap/6)
+exten => 6007,1,Dial(Zap/7)
+exten => 6008,1,Dial(Zap/8)
 
 ; Pre-configured SIP-phone extensions.  Primary use case is multiple SIP
 ; extensions and FXO analog Ports
 
-exten => 6011,n,Dial(SIP/6011)
-exten => 6012,n,Dial(SIP/6012)
-exten => 6013,n,Dial(SIP/6013)
-exten => 6014,n,Dial(SIP/6014)
-exten => 6015,n,Dial(SIP/6015)
-exten => 6016,n,Dial(SIP/6016)
-exten => 6017,n,Dial(SIP/6017)
-exten => 6018,n,Dial(SIP/6018)
-exten => 6019,n,Dial(SIP/6019)
-exten => 6020,n,Dial(SIP/6020)
-exten => 6021,n,Dial(SIP/6021)
-exten => 6022,n,Dial(SIP/6022)
-exten => 6023,n,Dial(SIP/6023)
-exten => 6024,n,Dial(SIP/6024)
-exten => 6025,n,Dial(SIP/6025)
-exten => 6026,n,Dial(SIP/6026)
-exten => 6027,n,Dial(SIP/6027)
-exten => 6028,n,Dial(SIP/6028)
-exten => 6029,n,Dial(SIP/6029)
-exten => 6020,n,Dial(SIP/6020)
+exten => 6011,1,Dial(SIP/6011)
+exten => 6012,1,Dial(SIP/6012)
+exten => 6013,1,Dial(SIP/6013)
+exten => 6014,1,Dial(SIP/6014)
+exten => 6015,1,Dial(SIP/6015)
+exten => 6016,1,Dial(SIP/6016)
+exten => 6017,1,Dial(SIP/6017)
+exten => 6018,1,Dial(SIP/6018)
+exten => 6019,1,Dial(SIP/6019)
+exten => 6020,1,Dial(SIP/6020)
+exten => 6021,1,Dial(SIP/6021)
+exten => 6022,1,Dial(SIP/6022)
+exten => 6023,1,Dial(SIP/6023)
+exten => 6024,1,Dial(SIP/6024)
+exten => 6025,1,Dial(SIP/6025)
+exten => 6026,1,Dial(SIP/6026)
+exten => 6027,1,Dial(SIP/6027)
+exten => 6028,1,Dial(SIP/6028)
+exten => 6029,1,Dial(SIP/6029)
+exten => 6030,1,Dial(SIP/6030)
 
diff --git a/easy-asterisk-gui/ipphone.png b/easy-asterisk-gui/ipphone.png
new file mode 100644
index 0000000000000000000000000000000000000000..f9578db1ea0dcf6f35dbea4c5a7dd6b5e998b5af
GIT binary patch
literal 2177
zcmV-{2!8j8P)<h;3K|Lk000e1NJLTq002Ay001Ef0ssI23f5m>00009a7bBm000XU
z000XU0RWnu7ytkO2XskIMF-je1p*))fXUE)000OeNkl<ZSi{YjYj6|S701urrzAi0
zFtRO-jfrC$?BHMmh8XiG)J#JOW=NQpB%RC@($Y>koj#^zrk!a%luny=G9+my3C*<m
zKuQ86DX&7X2Mi`H*p4k>8)NxxBg>NYT1mTF?Xw>YMwZr~0LkQhyLa#I{oS+o{Li@;
zsj3QTTU3yb_6O~LDXE{vpqPq1z1UWmt%EP$i1dfDqWu1iz~Nz8P-gWbfYnCsk0;L7
z?|Tux5=$3akd=>}fg>Zy!~pPQj(qO?#c^M7G8{8dBmhttSPc{aNQ&y=#D|D$ATYCz
zNDEq0)TYtUdxLUJPJXmzp7P<z<Kw;{0Q}SLx}i#01iVy+KU;1EfX1=NemkeA4~Lk<
zU~d_|Yl$^2=$3KjK#!10ywn7a&MVWzw|5F`wyC~=AHeDi@_)O|sVX!S;bt1UZ`P<F
zpBPsJ1y)(q3`|OEdpJCiLxX%YIrhM&2jscYXz!hFZdzKFZOMHS!R61}7tJ$8h0;HD
zGJ!a3EhL^RCK?+X)8Ru1eXG9SW-%rb?{D`B3KZyp_YO{yRsh>QpvS}fp?<cPN$$@k
z;ZfOj;QW>8W~t3uwY+wegPLr<IV_YG84oyuy^LC6L_e%F?tS?u$Nuq7I?$VUd}r4)
zJ7$_HK+`DmnODUSEVsaa`hvyt3+z)Ix{Kt6<I)>;-<b(W8llHwgE8{@$EO1kw)&9;
zlcC6-?c18BsH9;g3SO@@p7`XWU%m9AqA2$yo?raP|Jpl>a_+d<8DRIdMkZskqsSa5
z?h<_e9%LFPA$3jFM(N<GbN&dsx_r^jwKd}r@q#az7)S7nrF!@1@b7;1ayrBmO~3Zm
z!9-#vF8#GTnK+LW+dEpdp;dsZ6XC-{qL?yWg%KJrHr@1xRy<l%U9-W?VIw?9DJ$Et
zkf<~f>$X16hM9CW#`k{w(`PnsoM|d5oEl;}0<*?o2%_UX?jWNLJrv~L>Wwie3TvW3
z^*XPOxC(Mi^=mgY`?NJ-wH58IFpao9um1TjX-dS#4eJOD%{05FqKAjXB&{rf;0e29
zVXj33+Q$ip`y)fqlstsNQo{FI(`OkvdhfQU&iP5r=@(`|LlG5-iP<L0vQ@QdLI+|}
zAST`3naaDpY(SVbz)V3`+f`oFk~~FKPY#FAPH05u<0kdUnHE`9zPoAtWyT=Q5x_1E
zUa`jk;t!HX;wd=+dHZ_DT-s^8YbtttOiImR77CcL>m%MkLrv9`zA!vDlB9wpKjdPg
zh~gVNJG5($5HG&h$9egbRRWaa{)7MRD9AAtl`m@#-T(9-)X($#ruKZ&y?)h-ebw1_
z<W*3Q4zedlW187}thX%?<z{g1M3StiAN7Y?eHwp#l|??@+``2Lj*oLH`5->C>TnC<
zrI3flQLY}3dT(}jwsm*56&hgQa$`*vpZbbTa9`<lCTbgzpyb@Cn{q0$nS$|t$FSEA
z0EP3+lA>9BK@~pb_603j20cx>e2mqYiDTGgFs!G^{K7@eo`g9Q<~Y+m_P$UG01cgi
zU)SVTvVE=a_$*>dY;L)nlmKseIxZ=vMp%!=I$^OscCxJt01Pzs?X`9N5sl4LYZ5ND
zb#^%i3`o4tX76$iWMI<yw$3i+fGP=}2kupJ09Y~&0Kf-4Nf|0Qa?KM8YuD3mHco)J
z)u!>qR>$?YC;`B;waXo`tkhpoktLlz-yG&*->52U=@|-h{5Pu?H`)8RxKL4KL;2;A
za8gwSGlIOcZrMgfrzcQx79on_1y>AA35i|tA9r-8u93LCCiH5XSJMr(IvN1V=I3TC
zsOS%A)Oy)bSH3wb-<(yPYjOEQd8SO8IXl;wnQP1}SXAx^-_6G=&x)SGU^IGRbo|Uf
z&|UV_%zAQYa9p_>m!vsQE+Jl;^K7HfB5{06U2Qvql8=TsVD9U0UTURDV%Pd7UisG<
zJx%VctN#55p8`O`x>Y@*@$LtJV{Ohuv0?xVkj3aR0wru1+1=ZB)15S4Pm#abww9Pq
zD3@JxIz*>8Y+Jp|Dd(gHX-y_KIKgdMSy5y)#{_Z9%8EsKISj{dSy`c@Y1y_!jA?mS
zO(?<;LIkBH%(1if&Q#)eb~n7RqMQJL3**tQfQTV&XdY4dB~=~4<o+cY;@^)QXSi5p
zk&R*bOFaY2OA5WA$W`YkiQ_xAZ6Ax>r!3MK4EMB#*jexEDDv&^H@vW-93em+R`}7S
zfU2r!4rmI4F*dnCHB+$GI_IojPHYX4nn%MDel#BP4dYfzUO}$EcS=Cvyvn383k28E
zPtVS-@6g-38}`&zPBWt4wv4uV?|on{$Rq<IQUOO8#!A3S4=qLOaF|Dfl?FE}>2cLU
zNJmbe^fI%Q^_N@M?Qd8^;kN{uI6oE{VuX8s`kLpKaOwzbeg;WnVAhdg5v+P5B4a9m
zErSdS+CK|NU?i0ZRssN<4klPpkW~N>Wm#5~pePT-<mqFF3b0ToB@!cqe!itHk@$;U
z)&PL40xK&0q3Lod4qviO6aQg*T_W+FjQI7m=hrMtCH`7N*H@Lj@mzf>@z)o+zN+l-
z4;sum@<A^kzcSmjwDiNh+vg$%0Q9b7GAt<Re9PfwSy6FC#i%F;<BIf<uIOkS<!2Bj
zMj8O-XHt*l&`+0GEC%Xs0s4jRrzJ&|6_t}@Urb^Jc_75IqB6|HgK>HKa%Y%{izyeu
zB__&r+ig$e88CzvW|25TnG{arSO$gX8c5B(JD2?*7<R;5I`|SY00000NkvXXu0mjf
DlK2s5

literal 0
HcmV?d00001

diff --git a/easy-asterisk-gui/menu.html b/easy-asterisk-gui/menu.html
index c4e6cc99..4eb1af07 100644
--- a/easy-asterisk-gui/menu.html
+++ b/easy-asterisk-gui/menu.html
@@ -1,4 +1,4 @@
-    <div id="admin_tip" class="tip">Change your password, Reset the defaults, Install new software</div>
+    <div id="admin_tip" class="tip">Change your password, Time & time zone, Reset the defaults, Install new software</div>
     <div id="dashboard_tip" class="tip">Monitor the status of your phone system</div>
     <div id="network_tip" class="tip">Connect the phone system to your network and the Internet</div>
     <div id="phone_tip" class="tip">Set up your phones and phone calls</div>
diff --git a/easy-asterisk-gui/network.sh b/easy-asterisk-gui/network.sh
index d8172a19..d57854cb 100644
--- a/easy-asterisk-gui/network.sh
+++ b/easy-asterisk-gui/network.sh
@@ -66,7 +66,7 @@ cat << EOF
 EOF
 
 cat tooltips.html
-echo "<table align="center" width=800 border=0>"
+echo '<table align="center" width=800 border=0>'
 cat banner.html
 echo "    <tr>"
 cat menu.html    
diff --git a/easy-asterisk-gui/phone.pl b/easy-asterisk-gui/phone.pl
new file mode 100755
index 00000000..f540360e
--- /dev/null
+++ b/easy-asterisk-gui/phone.pl
@@ -0,0 +1,121 @@
+#!/sbin/microperl
+# phone.pl
+# David Rowe 5 Jan 2010
+#
+# Extracts the phone extension infor from /etc/asterisk/extensions.conf
+# and generate html.  Perl used as faster than equivalent shell script
+
+$tool_tip = "onMouseOver=\"popUp(event,'network_internet')\" onmouseout=\"popUp(event,'network_internet')\"";
+
+my %analog = (); # analog extension keyed on zap port
+
+open EXT, "/etc/asterisk/extensions.conf";
+while (<EXT>) { 
+    if (/.*=>[ ]*([0-9]*),1.*Zap\/([0-9]*)\)/) {
+        $analog{$2} = $1;
+	#print "'$1' '$2' $analog{$2}\n";
+    }
+}
+close EXT;
+
+my %zap = (); # zaptel port type keyed on zap port 
+              # (fxs/fxo or no entry if not live)
+open ZAP, "/etc/zaptel.conf";
+while (<ZAP>) { 
+    if (/fxoks=([0-9]*)/) {
+        $zap{$1} = "fxs";
+    }
+    if (/fxsks=([0-9]*)/) {
+        $zap{$1} = "fxo";
+    }
+}
+close ZAP;
+
+my %ip = (); # ip extension keyed on sip.conf name
+
+open EXT, "/etc/asterisk/extensions.conf";
+while (<EXT>) { 
+    if (/.*=>[ ]*([0-9]*),1.*Sip\/([0-9]*)\)/) {
+        $ip{$2} = $1;
+	#print "'$1' '$2' $ip{$2}\n";
+    }
+}
+close EXT;
+
+my %sip = ();  # SIP IP phone status keyed on sip.conf names 
+               # if no entry we can't see IP phone device
+my %voip = (); # SIP trunks status keyed on sip.conf names 
+               # if no entry we can't see SIP trunk
+my %ipad = (); # IP address of SIP device keyed on sip.conf names
+
+open SIP, "sipshowpeers.txt";
+while (<SIP>) { 
+    if (/^([0-9]*)[\s\/].*(OK)/) {
+        $sip{$1} = $2;
+	#print "'$1' '$2' $sip{$1}\n";
+	$e = $1;
+	if (/\s([0-9]+\.[0-9]+\.[0-9]+\.[0-9]+)\s/) {
+	    $ipad{$e} = $1;
+	    #print "'$1'\n";
+	}
+    }
+    if (/^(voip[0-9]*)[\s\/].*(OK)/) {
+        $voip{$1} = $2;
+	#print "'$1' '$2' $voip{$1}\n";
+	$e = $1;
+	if (/\s([0-9]+\.[0-9]+\.[0-9]+\.[0-9]+)\s/) {
+	    $ipad{$e} = $1;
+	    #print "'$1'\n";
+	}
+    }
+}
+
+close SIP;
+
+# print list of analog phones
+
+$tool_tip = "onMouseOver=\"popUp(event,'phone_phone')\" onmouseout=\"popUp(event,'phone_phone')\"";
+
+foreach $a (sort keys %analog) {
+    if ($zap{$a} eq "fxs") {
+	$icon = "<img $tool_tip src=\"phone.png\" alt=\"Analog Phone\" />";
+	print "<tr><td>$analog{$a}</td><td>Port $a</td><td>$icon</td></tr>\n";
+    }
+}
+
+# print list of IP phones
+
+$tool_tip = "onMouseOver=\"popUp(event,'phone_ipphone')\" onmouseout=\"popUp(event,'phone_ipphone')\"";
+
+foreach $s (sort keys %sip) {
+    if ($sip{$s} eq "OK") {
+	$icon = "<img $tool_tip src=\"ipphone.png\" alt=\"IP Phone\" />";
+	print "<tr><td>$s</td><td>$ipad{$s}</td><td>$icon</td></tr>\n";
+    }
+}
+
+print '<tr><td colspan="2" align="left" valign="top" ><h2>Phone Lines</h2></td></tr>
+';
+
+# print list of analog phone lines
+
+$tool_tip = "onMouseOver=\"popUp(event,'phone_phoneline')\" onmouseout=\"popUp(event,'phone_phoneline')\"";
+
+foreach $a (sort keys %analog) {
+    if ($zap{$a} eq "fxo") {
+	$icon = "<img $tool_tip src=\"phoneline.jpg\" alt=\"Phone Line\" />";
+	print "<tr><td>$analog{$a}</td><td>Port $a</td><td>$icon</td></tr>\n";
+    }
+}
+
+# print list of SIP VOIP trunks
+
+$tool_tip = "onMouseOver=\"popUp(event,'phone_voipline')\" onmouseout=\"popUp(event,'phone_voipline')\"";
+
+foreach $s (sort keys %voip) {
+    if ($voip{$s} eq "OK") {
+	$icon = "<img $tool_tip src=\"voipline.jpg\" alt=\"VOIP Line\" />";
+	print "<tr><td>$s</td><td>$ipad{$s}</td><td>$icon</td></tr>\n";
+    }
+}
+
diff --git a/easy-asterisk-gui/phone.png b/easy-asterisk-gui/phone.png
new file mode 100644
index 0000000000000000000000000000000000000000..9783372f330515e3433a32bb6ccca158c48e87a5
GIT binary patch
literal 2314
zcmV+l3HA1gP)<h;3K|Lk000e1NJLTq002Ay001or0ssI2eH5P?00009a7bBm000XU
z000XU0RWnu7ytkO2XskIMF-je1p*!_d9iK`000Q7Nkl<ZSi{Yk*>e-g9mjuj=)RAU
zWLdIhgRxm|3~Pu(ZNYAqs_b8oR4P@0t-vNvz%KhnRbE1pr{u5LmsB=MC4t2p2HRM^
zv1LoPC10|wTb3+Y*Jx%s4`aqcvauP%LO*r4db&UQ&F}cOgdv3BZMTPA-%BO5My;uu
z_xikJV*?D!{`GJF*w6m>TPOB<M-?)*mUw1ndU&{BE|&rVmSsO6Xm|gUcw|<oR94NW
zQYo)*T&0o&fLt#1ct-Yf3Mu-@QmL5Du2;^4!^@!+Z&OoK)yi}_^`!d;gf^{95O`)i
zl}JQIAB~Cj>HGJ-!%#^WW-=L4Lb@7VwKjhI<URmbIjK<S&$OQvckZLd4@FvfVtha*
zL3+J~A10GYoWN^|*Vfkl^S^(ol+wycmgVeryTxJ=aqi0MN;;hs<I^-vF9*GPy`CR3
z46_<qZE{t61<SIJpFF4{1^}f}d-TXL5$AFoH!|XuD`aAzyB_@AUT5X|<#IVN7dU?G
zIEG=BYZ!)k^!NdWu{DvvUb^@*k`(iX2Z!7=T@;f&<^nzvr-fl45U8~=0CYcnoXxIP
zP3Cfi%a^~X%`JOr$+e{4@56CS4D`rIx8RP;%gZ*KO(v7om}GEmEn4F(Tt~+RlhGtD
ziu(KeluEgnJpHiipUo}@-;YM45~;*sFx1A%(vqJbaKTUV`4U3NXtcE+Y84sx6XUsj
zT1;_?$HN%p34#ECTrRhsS+`kjRcrG3{LoN8{&H7@z+kX8H66ro*`<r0iX+bDa)CJ?
zNf2U+OK;CTSAz`z5JJIVu(fp?v-S2pW7$$gP?Dd0`gwXi-G2Hkj*G0m?t$k*VR|!Y
zuXof+>G-~9(sQ8sKrKYS`*|uA6*2=!$}U~JjN`b`U=&^aLg7$69+F7NH`X4ROxiy)
zqEILRAQ%ifolccXRkdbmX)zcK2*kxA-G2I<Mx)t}#we9aZg+1L@m`@{jP*4*>j5B<
zNXX@KgT6ZWve~SAu$Lq^-L6=o+fSc!I2`+wUq0Uy$I)U`-r~Y6$7J|Al1is2imF<`
zFwC=_CnSNr3_f(Qxvj13@IHyt>9lWpqN--?3A#|o4-RzcbUFZ-oSJNGXsq33aLAp{
zZ^=D^kau*P-zPD{Fx^idNu?4o(sW|n?P_!Yz{0|Uz0NL`Rtt%_`MJeKzf86{FO@1U
zfBr?~0)JpSyOAS_&4bh$-Ns8nEA-WBZA;4`mML#-I~Sno98r<x_6nU$#=@(!jg1Yw
z`mtKAwQGil`vu~1nY(bIv+|1-3i)EG90@NM3ONAK>&@|aIGs)cfJ&`Zrd9EH7ywS3
zJjYesWs1e(@TglZlZd&wJnjBBPrMphwcG8r3p}0)lHBqZoz7sPUJ;zr)06r922YG(
zm`-oxi2;D3>Ud(gT-kE46`?J`-P8M2CMCt(T*gNG%_c1$ScO8N-d<npcK(3xwULy<
z328h$FO|swz){~64hQ)@Mb)KK3BKRi`5EFE;i=$qFqw?hrn%jr^ZAWA|LB1OO#l#%
zgzIc|+vLF)V=PMxB?cj6v)L=f^vdcA$8jd31ppLEwNxrI7|guMtyV{b3o4arhqIZY
zs4`s?gs$7&^LpYPp@#;$o0~VUA^BY1UT3e>7=ig|3AyE6Bq=|6;?!2H3C<Uah2WB(
zA2zxU&ChxHt6f85%Zsr-j^hY|Jawu=AO--><OGIsyXLfQKAxSOV(2uV+Crhw_TD@>
zE+l3E$mfe^+B<}7NYiwoP>M!Y_@Ti_rBex>ST0x4bcy3Q0BCJHu2jf{O%jR3%*^Co
zBnnw`l`1XHk5gtN|BBaAYqikDVqsyyFOd*}V>G#191dqi9`xflS!Ztm03657CR;2X
zG8icUuv#6_=!&bURiRKFJ9>&K7q_lbqk~fE?pmnj;F0P@eSKX9y_)aOFU)5*vYshV
z)f&HlRv-od43nHc-&tX>7+{2VY7_tgfX(KdSRE)9^8jEpTGo=WOeO^YpM3IjffxY1
zzNz)~#7>D>Xd12gSHHeVRtckGKCahT1rxfOUA0fHt}f>`1Ph__=R1YNSe6M#!l6)*
zCzeR01VQq|7=~*!x}_yQ09dT`7K-9tCJmaVeLl~w#A=QC+Vwkny@{-h_vgL$Y}-|k
zs-`y>a~mnXZ?`u%>bHbZEFRP8DY{%XQ+2#E#G<RNrd9y3Sn3n;FyB9U;;cZ75bAs0
zgX3s7V$=1n?&$TVSA2SC=$Tx$Ct-j9u`EjvM4?EZY451~%_$OzEH2LQab_}ElZj|H
zn+5=-N|VoJW3dnb965T1Br(KoQZ6hkgu_9htZv&>YU6LOZ4z%~k5Z|aj4$%SXxHX+
zItzso%d#Jz?X0}jMq|-bI>8eY1ff)EcwzvsSnNEpTCHts{RnZK@Omdaqq`D+^}9b9
zjpkRf2SVs!*Y`FnwRaVU5Yp+WLoJ6Yeo0{&7UgnTzRuZf&eU3r?;8!&Ms}U=A3y#v
z1{~T_hy!%FK<tp_5CRNWT)TcpuQyhwyro4yiI;F(w?|?e$7OO==TClGIY|)Y;kF}(
z+m5{c*>btWv5cUY=jNsrik+Pjli#@cN29UYni#{BAKd@WW;Jbh%gRKek!^R=YPGdj
z@i%G+q5D7lw^E2#eTqQ-^*4X6B?bVp=h=OuL9-K?R;z_~(<Y}pnM`u0-^UP=-MDpk
zTVenphlih<%m(<;+l+5&Ct?msZ+-I@lWE5oBh^}^)n@(ip{G1ANK&#b@r|2zUqcM<
z;y;)DG>((rxOtbNZ13vti?`bl0uD)Tef=l1dH2`;p+Gk@e)Nl9{$8uq?`1Lhfe>H)
z@>_z~E6n@%X=lUX-(0!9cjEUC?QA%F<?3yTMDj*Yy*KwjfE>=Yt6zS*ZQ|PnIv$I^
k9~(Q{u3Wt>f*1h)4{U7gH-Q23D*ylh07*qoM6N<$g5VToUjP6A

literal 0
HcmV?d00001

diff --git a/easy-asterisk-gui/phone.sh b/easy-asterisk-gui/phone.sh
index 1d3a5de9..dc84c546 100644
--- a/easy-asterisk-gui/phone.sh
+++ b/easy-asterisk-gui/phone.sh
@@ -23,30 +23,27 @@ cat << EOF
 <html>
 <title>Easy Asterisk - Phones</title>
 <body onload="localInit()">
-<div id="t1" class="tip">Tells you if I can reach the Internet.  If 
-                         not "Good" check your network settings, in 
-                         particular Gateway and DNS.</div>
-
-<table align="center" width=800>
 EOF
+
+cat tooltips.html
+echo '<table align="center" width=800>'
 cat banner.html
 echo "    <tr>"
 cat menu.html    
 cat <<EOF
 
-    <td>
+    <td valign="top">
 
-    <form action="/cgi-bin/set_phone.sh" onsubmit="return validate_form(this)" method="get">
     <table align="right" width=600>
-    <tr>
-      <tr><td colspan="2" align="left" valign="top" ><h2>Phones</h2></td>
-      <tr onMouseOver="popUp(event,'t1')" onmouseout="popUp(event,'t1')">
-	  <td>Internet Connection:</td>
-	  <td><div id="internet" >
-	  <span style="margin-left: 4px;font-weight:bold">&nbsp;</span></div></td>
-      </tr>
+      <tr><td colspan="2" align="left" valign="top" ><h2>Phones</h2></td></tr>
+EOF
+
+# use perl to construct list of phones and phone lines for us
+asterisk "-rx sip show peers" 2>/dev/null > sipshowpeers.txt
+./phone.pl
+
+cat<<EOF
     </table>
-    </form>
 
     </td>
 
diff --git a/easy-asterisk-gui/phoneline.jpg b/easy-asterisk-gui/phoneline.jpg
new file mode 100644
index 0000000000000000000000000000000000000000..21078c3e7883db899104c50b1198aca22871869d
GIT binary patch
literal 1198
zcmex=<NpH&0WUXCHwH#VMg|WcWcdG(LCm!xGmU}4*Ox(yfq{Voh=Jh$Z3brsR%R9!
z7G_o;!OF_Y#?HgR4g~z%+?+gu{6a#4{DOkQVlv{wB2uD)f)a`nQnIr0^76vsN-9cn
zDl&5Nav(z(fm+$w*!eg(_~b+cMdU~Z{|_(-axj=N*f29HF)#@-G7B>PKf)l(z`(@B
z4DtpXu(NS6v9NM7G6SWJ1sE8anVEntXJLmZU}R!uVPF+xQ)Cx1bW{pVY~&CYF;aFa
zEJ|*gxNzgek3dZlj0`{%;pTxDAYDKiX+fY(Kq+Q~BqNicA+wO;|62?^%#1*HF$*%-
zGh`IlKe9TyIy&j%YHqRYlZS-96-qhn<@_yD(|6B(tKC7C6q)8YKlj?))e5sJKSV!b
z56;-KId|UTs!hu|1?Lz1Ip+T1TDkm|CzZ)-750gQn78lozGU@^{rtw5Nh>+sjW^8w
z;F15cx<7E@>!9eA<!T#Lm-@C%vC)t(_<MZgm!k%E7j4MAVtGbrftR0%b@P*KD~3f~
z8eIIx;uzw;RKGj3ecQ@uZzfF-F!T8O=|6+Ww91tR_cThmbCWkptPfvebfqoi^xLJ!
zt{5y?ks|1^Pxp^qq$!u|?1bDMMSUTW${()O+U#Ay6`Z#JRHN$lTcyY1aveQNmx`=p
zKO?xqerd!z!NRhaTAG_*?OxLHX3rAM*|X;8T3$)ec{}fj&fA-jYzwFPNf<8E|1{(3
zw3EI~OH|trJa>*>t;H;-*C~0U=XuqiukF0+%&z{8T@|qF@T8(klh&1Ueq#SD@A&BT
zLy;YII%hXW8T0+iF}@$A$Ler+u8jDEUw$6W2U3<su>9C|Kt*b9q$y*mR_w`Z(W>Vh
zgeD2iSDw6+ZBkipu;T3EJhcP==I_$9H;d+T*Ih5OgHK^jv3Zl5`|W*hKW(jb54j#b
zCKIn<r>gNp_Cw+Ox54+SE<OLe$L`XSm5QE<p2`Y+ZTl`Yv8|e#va@E!B)esnj9#gk
zH~OC1zu#*vQDAjSR+@E*M)lQA>8>IHf^zB+5qoY)KFbig{4#0^+mC}1|GsSAmT4OG
z+_<n|U)hBhMv0nn;sX2yZpU{hTO4?KZ9bnOpTfdSE>8)C?<@YkJNEsO)$Dq$3tQ%F
zdG<v3uKKx2PkTjYd{X+<vMR<T)#c6d2-||f%j;h@UYqiLw{dV}?Xj7YbDq?!pCW&_
z(DJ4$^Wk|v8LBTc-Fp%pecoDi>xL&&{%jAKx0qRa`8vZp_2*KyEo?ElQaokbMW^%j
zMzQzTFHQgUcIUeO{&2CZu%(M+EVdnN^Pbbmb-z?Q@Az$#a~CJgoxbeRzK<U#EZm<U
z;vj!nlznSN-gYH!R-NrXr|oHMGCv@{a+&Qq=>i2)!97LW=HwV?eL7t8$t+`0*RrH8
zjRlt5ck5_xb~F-fU0$-**0pGDRpc}k&7iRM25W{nyt`*<EKP~aedKjHBa@MLiK_Af
NrcH~wGV1@|1OROO$|e8+

literal 0
HcmV?d00001

diff --git a/easy-asterisk-gui/sip.conf b/easy-asterisk-gui/sip.conf
new file mode 100644
index 00000000..3cd56ed3
--- /dev/null
+++ b/easy-asterisk-gui/sip.conf
@@ -0,0 +1,546 @@
+;
+; SIP Configuration example for Asterisk
+;
+; Syntax for specifying a SIP device in extensions.conf is
+; SIP/devicename where devicename is defined in a section below.
+;
+; You may also use 
+; SIP/username@domain to call any SIP user on the Internet
+; (Don't forget to enable DNS SRV records if you want to use this)
+; 
+; If you define a SIP proxy as a peer below, you may call
+; SIP/proxyhostname/user or SIP/user@proxyhostname 
+; where the proxyhostname is defined in a section below 
+; 
+; Useful CLI commands to check peers/users:
+;   sip show peers		Show all SIP peers (including friends)
+;   sip show users		Show all SIP users (including friends)
+;   sip show registry		Show status of hosts we register with
+;
+;   sip debug			Show all SIP messages
+;
+;   reload chan_sip.so		Reload configuration file
+;				Active SIP peers will not be reconfigured
+;
+
+[general]
+context=default			; Default context for incoming calls
+;allowguest=no			; Allow or reject guest calls (default is yes, this can also be set to 'osp'
+				; if asterisk was compiled with OSP support.
+;realm=mydomain.tld		; Realm for digest authentication
+				; defaults to "asterisk"
+				; Realms MUST be globally unique according to RFC 3261
+				; Set this to your host name or domain name
+bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
+bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
+srvlookup=yes			; Enable DNS SRV lookups on outbound calls
+				; Note: Asterisk only uses the first host 
+				; in SRV records
+				; Disabling DNS SRV lookups disables the 
+				; 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")
+;tos=184			; Set IP QoS to either a keyword or numeric val
+;tos=lowdelay			; lowdelay,throughput,reliability,mincost,none
+;maxexpiry=3600			; Max length of incoming registration we allow
+;defaultexpiry=120		; Default length of incoming/outoing registration
+;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
+;checkmwi=10			; Default time between mailbox checks for peers
+;vmexten=voicemail      ; dialplan extension to reach mailbox sets the 
+						; Message-Account in the MWI notify message 
+						; defaults to "asterisk"
+;videosupport=yes		; Turn on support for SIP video
+;recordhistory=yes		; Record SIP history by default 
+				; (see sip history / sip no history)
+
+;disallow=all			; First disallow all codecs
+;allow=ulaw			; Allow codecs in order of preference
+;allow=ilbc			; 
+;musicclass=default		; Sets the default music on hold class for all SIP calls
+				; This may also be set for individual users/peers
+;language=en			; Default language setting for all users/peers
+				; This may also be set for individual users/peers
+;relaxdtmf=yes			; Relax dtmf handling
+;rtptimeout=60			; Terminate call if 60 seconds of no RTP activity
+				; when we're not on hold
+;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP activity
+				; when we're on hold (must be > rtptimeout)
+;trustrpid = no			; If Remote-Party-ID should be trusted
+;sendrpid = yes			; If Remote-Party-ID should be sent
+;progressinband=never		; If we should generate in-band ringing always
+				; use 'never' to never use in-band signalling, even in cases
+				; where some buggy devices might not render it
+;useragent=Asterisk PBX		; Allows you to change the user agent string
+;promiscredir = no      	; If yes, allows 302 or REDIR to non-local SIP address
+	                       	; Note that promiscredir when redirects are made to the
+       	                	; local system will cause loops since SIP is incapable
+       	                	; of performing a "hairpin" call.
+;usereqphone = no		; If yes, ";user=phone" is added to uri that contains
+				; a valid phone number
+;dtmfmode = rfc2833		; Set default dtmfmode for sending DTMF. Default: rfc2833
+				; Other options: 
+				; info : SIP INFO messages
+				; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
+				; auto : Use rfc2833 if offered, inband otherwise
+
+;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 create and destroy a
+; NoOp priority 1 extension for a given peer who registers or unregisters with
+; us.  The actual extension is the 'regexten' parameter of the registering
+; peer or its name if 'regexten' is not provided.  More than one regexten may
+; be supplied if they are separated by '&'.  Patterns may be used in regexten.
+;
+;regcontext=sipregistrations
+;
+; Asterisk can register as a SIP user agent to a SIP proxy (provider)
+; Format for the register statement is:
+;       register => user[:secret[:authuser]]@host[:port][/extension]
+;
+; If no extension is given, the 's' extension is used. The extension needs to
+; be defined in extensions.conf to be able to accept calls from this SIP proxy
+; (provider).
+;
+; host is either a host name defined in DNS or the name of a section defined
+; below.
+;
+; Examples:
+;
+;register => 1234:password@mysipprovider.com	
+;
+;     This will pass incoming calls to the 's' extension
+;
+;
+;register => 2345:password@sip_proxy/1234
+;
+;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider
+;    connect to local extension 1234 in extensions.conf, default context,
+;    unless you configure a [sip_proxy] section below, and configure a
+;    context.
+;    Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
+;    Tip 2: Use separate type=peer and type=user sections for SIP providers
+;           (instead of type=friend) if you have calls in both directions
+  
+;registertimeout=20		; retry registration calls every 20 seconds (default)
+;registerattempts=10		; Number of registration attempts before we give up
+				; 0 = continue forever, hammering the other server until it 
+				; accepts the registration
+				; Default is 0 tries, continue forever
+;callevents=no			; generate manager events when sip ua performs events (e.g. hold)
+
+;----------------------------------------- NAT SUPPORT ------------------------
+; The externip, externhost and localnet settings are used if you use Asterisk
+; behind a NAT device to communicate with services on the outside.
+
+;externip = 200.201.202.203	; Address that we're going to put in outbound SIP messages
+				; if we're behind a NAT
+
+				; The externip and localnet is used
+				; when registering and communicating with other proxies
+				; that we're registered with
+;externhost=foo.dyndns.net	; Alternatively you can specify an 
+				; external host, and Asterisk will 
+				; perform DNS queries periodically.  Not
+				; recommended for production 
+				; environments!  Use externip instead
+;externrefresh=10		; How often to refresh externhost if 
+				; used
+				; You may add multiple local networks.  A reasonable set of defaults
+				; are:
+;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
+;localnet=10.0.0.0/255.0.0.0	; Also RFC1918
+;localnet=172.16.0.0/12		; Another RFC1918 with CIDR notation
+;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
+
+; The nat= setting is used when Asterisk is on a public IP, communicating with
+; devices hidden behind a NAT device (broadband router).  If you have one-way
+; audio problems, you usually have problems with your NAT configuration or your
+; firewall's support of SIP+RTP ports.  You configure Asterisk choice of RTP
+; ports for incoming audio in rtp.conf
+;
+;nat=no				; Global NAT settings  (Affects all peers and users)
+                                ; yes = Always ignore info and assume NAT
+                                ; no = Use NAT mode only according to RFC3581 
+                                ; never = Never attempt NAT mode or RFC3581 support
+				; route = Assume NAT, don't send rport 
+				; (work around more UNIDEN bugs)
+
+;rtcachefriends=yes		; Cache realtime friends by adding them to the internal list
+				; just like friends added from the config file only on a
+				; as-needed basis? (yes|no)
+
+;rtupdate=yes			; Send registry updates to database using realtime? (yes|no)
+				; If set to yes, when a SIP UA registers successfully, the ip address,
+				; the origination port, the registration period, and the username of
+				; the UA will be set to database via realtime. If not present, defaults to 'yes'.
+
+;rtautoclear=yes		; Auto-Expire friends created on the fly on the same schedule
+				; as if it had just registered? (yes|no|<seconds>)
+				; If set to yes, when the registration expires, the friend will vanish from
+				; the configuration until requested again. If set to an integer,
+				; friends expire within this number of seconds instead of the
+				; registration interval.
+
+;ignoreregexpire=yes		; Enabling this setting has two functions:
+				;
+				; For non-realtime peers, when their registration expires, the information
+				; will _not_ be removed from memory or the Asterisk database; if you attempt
+				; to place a call to the peer, the existing information will be used in spite
+				; of it having expired
+				;
+				; For realtime peers, when the peer is retrieved from realtime storage,
+				; the registration information will be used regardless of whether
+				; it has expired or not; if it expires while the realtime peer is still in
+				; memory (due to caching or other reasons), the information will not be
+				; removed from realtime storage
+
+; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
+; domains, each of which can direct the call to a specific context if desired.
+; By default, all domains are accepted and sent to the default context or the
+; context associated with the user/peer placing the call.
+; Domains can be specified using:
+; domain=<domain>[,<context>]
+; Examples:
+; domain=myasterisk.dom
+; domain=customer.com,customer-context
+;
+; In addition, all the 'default' domains associated with a server should be
+; added if incoming request filtering is desired.
+; autodomain=yes
+;
+; To disallow requests for domains not serviced by this server:
+; allowexternaldomains=no
+
+; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
+                          ; non-peers, use your primary domain "identity"
+                          ; for From: headers instead of just your IP
+                          ; address. This is to be polite and
+                          ; it may be a mandatory requirement for some
+                          ; destinations which do not have a prior
+                          ; account relationship with your server. 
+
+[authentication]
+; Global credentials for outbound calls, i.e. when a proxy challenges your
+; Asterisk server for authentication. These credentials override
+; any credentials in peer/register definition if realm is matched.
+;
+; This way, Asterisk can authenticate for outbound calls to other
+; realms. We match realm on the proxy challenge and pick an set of 
+; credentials from this list
+; Syntax:
+;	auth = <user>:<secret>@<realm>
+;	auth = <user>#<md5secret>@<realm>
+; Example:
+;auth=mark:topsecret@digium.com
+; 
+; You may also add auth= statements to [peer] definitions 
+; Peer auth= override all other authentication settings if we match on realm
+
+;------------------------------------------------------------------------------
+; Users and peers have different settings available. Friends have all settings,
+; since a friend is both a peer and a user
+;
+; User config options:        Peer configuration:
+; --------------------        -------------------
+; context                     context
+; permit                      permit
+; deny                        deny
+; secret                      secret
+; md5secret                   md5secret
+; dtmfmode                    dtmfmode
+; canreinvite                 canreinvite
+; nat                         nat
+; callgroup                   callgroup
+; pickupgroup                 pickupgroup
+; language                    language
+; allow                       allow
+; disallow                    disallow
+; insecure                    insecure
+; trustrpid                   trustrpid
+; progressinband              progressinband
+; promiscredir                promiscredir
+; useclientcode               useclientcode
+; accountcode                 accountcode
+; setvar                      setvar
+; callerid		      callerid
+; amaflags		      amaflags
+; call-limit		      call-limit
+; restrictcid		      restrictcid
+; subscribecontext	      subscribecontext
+; videosupport		      videosupport
+;                             mailbox
+;                             username
+;                             template
+;                             fromdomain
+;                             regexten
+;                             fromuser
+;                             host
+;                             port
+;                             qualify
+;                             defaultip
+;                             rtptimeout
+;                             rtpholdtimeout
+;                             sendrpid
+
+;[sip_proxy]
+; For incoming calls only. Example: FWD (Free World Dialup)
+; We match on IP address of the proxy for incoming calls 
+; since we can not match on username (caller id)
+;type=peer
+;context=from-fwd
+;host=fwd.pulver.com
+
+;[sip_proxy-out]
+;type=peer          		; we only want to call out, not be called
+;secret=guessit
+;username=yourusername		; Authentication user for outbound proxies
+;fromuser=yourusername		; Many SIP providers require this!
+;fromdomain=provider.sip.domain	
+;host=box.provider.com
+;usereqphone=yes		; This provider requires ";user=phone" on URI
+;call-limit=5			; permit only 5 simultaneous outgoing calls to this peer
+
+;------------------------------------------------------------------------------
+; Definitions of locally connected SIP phones
+;
+; type = user	a device that authenticates to us by "from" field to place calls
+; type = peer	a device we place calls to or that calls us and we match by host
+; type = friend two configurations (peer+user) in one
+;
+; For local phones, type=friend works most of the time
+;
+; If you have one-way audio, you propably have NAT problems. 
+; If Asterisk is on a public IP, and the phone is inside of a NAT device
+; you will need to configure nat option for those phones.
+; Also, turn on qualify=yes to keep the nat session open
+
+;[grandstream1]
+;type=friend 			
+;context=from-sip		; Where to start in the dialplan when this phone calls
+;callerid=John Doe <1234>	; Full caller ID, to override the phones config
+;host=192.168.0.23		; we have a static but private IP address
+				; No registration allowed
+;nat=no				; there is not NAT between phone and Asterisk
+;canreinvite=yes		; allow RTP voice traffic to bypass Asterisk
+;dtmfmode=info			; either RFC2833 or INFO for the BudgeTone
+;call-limit=1			; permit only 1 outgoing call and 1 incoming call at a time
+				; from the phone to asterisk
+				; (1 for the explicit peer, 1 for the explicit user,
+				; remember that a friend equals 1 peer and 1 user in
+				; memory)
+;mailbox=1234@default		; mailbox 1234 in voicemail context "default"
+;disallow=all			; need to disallow=all before we can use allow=
+;allow=ulaw			; Note: In user sections the order of codecs
+				; listed with allow= does NOT matter!
+;allow=alaw
+;allow=g723.1			; Asterisk only supports g723.1 pass-thru!
+;allow=g729			; Pass-thru only unless g729 license obtained
+;astdb=chan2ext/SIP/grandstream1=1234	; ensures an astDB entry exists
+
+
+;[xlite1]
+; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
+; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
+;type=friend
+;regexten=1234			; When they register, create extension 1234
+;callerid="Jane Smith" <5678>
+;host=dynamic			; This device needs to register
+;nat=yes			; X-Lite is behind a NAT router
+;canreinvite=no			; Typically set to NO if behind NAT
+;disallow=all
+;allow=gsm			; GSM consumes far less bandwidth than ulaw
+;allow=ulaw
+;allow=alaw
+;mailbox=1234@default,1233@default	; 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
+;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
+;vmexten=voicemail      ; dialplan extension to reach mailbox 
+                        ; sets the Message-Account in the MWI notify message
+                        ; defaults to global vmexten which defaults to "asterisk"
+;restrictcid=yes		; To have the callerid restriced -> sent as ANI
+;disallow=all
+;allow=ulaw			; dtmfmode=inband only works with ulaw or alaw!
+
+
+;[polycom]
+;type=friend			; Friends place calls and receive calls
+;context=from-sip		; Context for incoming calls from this user
+;secret=blahpoly
+;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"
+
+
+;[pingtel]
+;type=friend
+;secret=blah
+;host=dynamic
+;insecure=port			; Allow matching of peer by IP address without matching port number
+;insecure=invite		; Do not require authentication of incoming INVITEs
+;insecure=port,invite		; (both)
+;qualify=1000			; Consider it down if it's 1 second to reply
+				; Helps with NAT session
+				; qualify=yes uses default value
+;callgroup=1,3-4		; We are in caller groups 1,3,4
+;pickupgroup=1,3-5		; We can do call pick-p for call group 1,3,4,5
+;defaultip=192.168.0.60		; IP address to use if peer has not registred
+
+;[cisco1]
+;type=friend
+;secret=blah
+;qualify=200			; Qualify peer is no more than 200ms away
+;nat=yes			; This phone may be natted
+				; Send SIP and RTP to the IP address that packet is 
+				; received from instead of trusting SIP headers 
+;host=dynamic			; This device registers with us
+;canreinvite=no			; Asterisk by default tries to redirect the
+				; RTP media stream (audio) to go directly from
+				; the caller to the callee.  Some devices do not
+				; support this (especially if one of them is 
+				; 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
+
+; Pre-configured SIP extensions
+
+[6011]
+type=friend
+context=default
+host=dynamic
+user=6011
+secret=6011
+canreinvite=no
+callerid=6011
+disallow=all
+allow=ulaw,g729
+qualify=yes
+
+[6012]
+type=friend
+context=default
+host=dynamic
+user=6012
+secret=6012
+canreinvite=no
+callerid=6012
+disallow=all
+allow=ulaw,g729
+
+[6013]
+type=friend
+context=default
+host=dynamic
+user=6013
+secret=6013
+canreinvite=no
+callerid=6013
+disallow=all
+allow=ulaw,g729
+
+[6014]
+type=friend
+context=default
+host=dynamic
+user=6014
+secret=6014
+canreinvite=no
+callerid=6014
+disallow=all
+allow=ulaw,g729
+
+[6015]
+type=friend
+context=default
+host=dynamic
+user=6015
+secret=6015
+canreinvite=no
+callerid=6015
+disallow=all
+allow=ulaw,g729
+
+[6016]
+type=friend
+context=default
+host=dynamic
+user=6016
+secret=6016
+canreinvite=no
+callerid=6016
+disallow=all
+allow=ulaw,g729
+
+[6017]
+type=friend
+context=default
+host=dynamic
+user=6017
+secret=6017
+canreinvite=no
+callerid=6017
+disallow=all
+allow=ulaw,g729
+
+[6018]
+type=friend
+context=default
+host=dynamic
+user=6018
+secret=6018
+canreinvite=no
+callerid=6018
+disallow=all
+allow=ulaw,g729
+
+; Pre-configured SIP trunks
+
+[voip1]
+type=friend
+context=default
+user=user
+secret=password
+host=192.168.1.28
+canreinvite=no
+disallow=all
+allow=ulaw,g729
+qualify=yes
+
diff --git a/easy-asterisk-gui/tooltips.html b/easy-asterisk-gui/tooltips.html
index f89cf56a..9e0795d0 100644
--- a/easy-asterisk-gui/tooltips.html
+++ b/easy-asterisk-gui/tooltips.html
@@ -8,3 +8,16 @@
                          Emergency backdoor IP. Useful if you get locked out of the main network connection, for 
                          example due to DHCP problems on your network or a configuration mistake.  
                          Write this number down somewhere!</div>
+
+<div id="phone_phone" class="tip">
+                         Analog Phone extension: Normal telephone plugged into a port on your phone system.
+                         </div>
+<div id="phone_ipphone" class="tip">
+                         IP Phone extension: IP Phone plugged into your network.
+                         </div>
+<div id="phone_line" class="tip">
+                         Analog Phone Line: Analog telephone line plugged into a port on your phone system.
+                         </div>
+<div id="phone_voipline" class="tip">
+                         VOIP  Phone Line: Make and receive phone calls over the Internet.
+                         </div>
diff --git a/easy-asterisk-gui/voipline.jpg b/easy-asterisk-gui/voipline.jpg
new file mode 100644
index 0000000000000000000000000000000000000000..d84b1052a3ea84483b694ca8b82fe0d47ef22930
GIT binary patch
literal 1809
zcmb7AcTkfF7XQ*is9zF-LI{F@z;Y^4AZiv2L8^c>Q2{{_oWP<IL19&jETUs5p#_jG
zkf6tE5Gg8B76=feWDTN3l)6%6gCI>ex$Mo&&D@{&d-L8uzw+ii`KbIOpyA@+>;OO@
z5Wq=MfP4ZV9EoLw0f3tuU<LpHTG0x~#{fEjfWhH#7(y`+2m}(Pf<`HzuB@!20;*%N
zpgIWR2t+L$URx6cx9M-w*45S1)5B>Q7#ffai6lMJ-ysl%6@^5pq0wq2JcuX#-;lQf
zAR3?oR2aknfPxSh2$6RJL;!$7VTvpM3j_*|guu`MRKZ$n01!9;hauqzDC*w|2owef
z0R&bbX+gsoc%F$$LfMt`caqr?QoQCq`?Fc?9U_HM`|rp6Z!8iDM?qi;ZZ`-(AutFM
zj)eVZts)eFgZc=p1<f-Gc_vAUXP5KaCvfDm!}524ib4hjVIW`&Op)D+2MlM1DhD(3
zLK1zP2vx_lCZld!_>r9>&6J$9^PcpYzDx+%E7OxMJPB*+zq(DyKD4%C-bbSfoAW0F
z;`bnfJ;yp1oo>Y*03IP-5isLwX@crG>0zdKWN+M#D|a({{249nRWMQ7@ey;ALU-|x
zSfAOkyGolXFYbP4-d@hKe!!?t@(;wdrUvW{mR@k%o@?8QebwZ8OWQor*{bULfEiD_
zd>)Cuzh|O55K$xYBXXs`YY>XuFu=Nl>`mg-7$Hsn{hiW56e)EL6>nLT=ap12tt4ZT
z53yJ`wnTM$YF+u?^dyv@t;1})RO=R~;-UK`qi;J)Q6t~I3ogB`J{fv6H5q55_ElVk
zDfe;G-I}#Fo_m-Vx@nl|!d#p@%5bP@?lnWM?Y!LbfzwEh94NfB5C<OT)p_mD5f$x^
z0l7C)OtM>NUsh*su&*AZWAn%Q!Bq;p#2(kSUbZwF;fS;z*KODH5Hsy>m85@T*`(Aj
zgqxnin^dL;v*O@y_NDrDz0`@IWEfnJU9!w~G&(hsc*qBOt#TvD_iA55hbQrI(a4H_
z@mmh+(={&whY@{GIDgQ0CAn^6xGvdJs=*QAr`|z(4tp*wq2WB^l#>r1TN5^#CsI33
z7O|T3-i`UE)u)hA7wrCNQWG#Je7fR>=~BYsOa!4^yA!g4`r{uXT%9rNhG2X#(3RA9
znn+r;>_!AuT>C!He$*lpW5&3*ENA&x`;=}bO+YIuJ?;CYFix|c^0_+4hVEr3R4)9I
zCym5U=J*6^J4~+IwA_ee3)^VASthHV64_|7>Y15x&KrAu1XsN;IxS->UXY;EbV)+$
zFR1f}jL3nV;l#mz#B&y%q$tfb^9Je7smJS!SIU=4W0<kNH}Q8KlkX$mVhJm4YwfPP
zp78CJrz0YkJ#{Vey^AB~866_DfrvfQxQgF<K;Y6|;cQlsP$CC}oxGaSvyCA?#ky$a
zBK^I~*12XFN2;4fOjmFHb6(T#*CwWeI+o2}TN^Y!XfV5(zy81>?W5%j^+!C1jOZLb
z|8wfFP8FSp`heM!=2p2Ja25HBW(Uo$V1%D=lo9(ptfRqne+a_e=4WsPnmwwex{Oe_
zSbNazRaujx?@NnL2>O7K#C1|Q?3$<b_4wr3Bsoy}>kkJm|CK$1`B7XtG$|yS(IbDk
z>GP=V7_9Kl>*S?)fyOZ99Q=!j#|<2LMG}@-ZG^jAVP(b`nLYI#h4qh;%1Y>3h#vF3
z634z%Y(9;6Pm{K*he=wGeK>u2XD9E0-D!Jl;F}xb7NG=sBzA7sEG4HvIo_@)=7=k`
z_qRi2!n2%<a=>@ukKbjbw2`D{M1HzmhQKbf0$*r*&_fbj=~eu^;@S9yW?qkbPWhOr
z$ILMaSJ)pHzjngP-|0j}a_`|$n^K`})bQdtjGZmFH6e@o@k@;mFw(WMoI9^%%OYpV
z0qtj^1R;6MaaAk_ezJ<aM@eJPTnNK9(w!>})}tJwE8|<QUw*xrF^ija&#LvXnJhZF
zV!UiR<@%K{7tTF?NYK!<DKpJbKUF%-$bgjbxCQO3Y|BNwe+<otF~o7-Qk7lwoJtCX
z4pKu7FH`<-no-O3`9usBPmq&U_V>{~>p}#9M9zdGZ)N!fUO29EF-5}IBM{|{2ZiN^
zFZMmJ+`1N=d|<I#{GPre8h>SLGCAzC{q2vn+oM9*<<%_fF4bkr0fW0){J)Miqa(BV
zF1KpW|K<_2(UCiD{K_W{9o6t+s94sR_-Cy^vYWb)Zoc9rK9707!7~pt$u|5t@vWA!
zwtG|9P~sDk7F!qu&Ke2u-rJv&x*jQSW)!p;GP|SIaT)bH`~n=@sM*a~^eo1wiW4~o
KDv%)g+kXLTs0ru*

literal 0
HcmV?d00001

-- 
2.25.1