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<DIV><FONT face=Arial size=2>Sorry if this comes in twice. Wasn't subscribed
first time :-(</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Anyone help me here......</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>It worked once :-(</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have a static IP address which is on my private
network.. Phone is 192.192.192.220 and asterisk server is
192.192.192.22</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have the 7690 with a SIP iamge (Whatever latest
is )</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have 3 lines setup with Free World Dial up and
have the 4th setup to connect to my asterisk server. Here are my config
files......It worked once but now the phone sits there with a 'x' next to it
:-(</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>;<BR>; SIP Configuration for Asterisk<BR>;<BR>;
Syntax for specifying a SIP device in extensions.conf is<BR>; SIP/devicename
where devicename is defined in a section below.<BR>;<BR>; You may also use <BR>;
<A href="mailto:SIP/username@domain">SIP/username@domain</A> to call any SIP
user on the Internet<BR>; (Don't forget to enable DNS SRV records if you want to
use this)<BR>; <BR>; If you define a SIP proxy as a peer below, you may
call<BR>; SIP/proxyhostname/user or <A
href="mailto:SIP/user@proxyhostname">SIP/user@proxyhostname</A> <BR>; where the
proxyhostname is defined in a section below <BR>; <BR>; Useful CLI commands to
check peers/users:<BR>; sip show peers Show all SIP peers
(including friends)<BR>; sip show users Show all SIP
users (including friends)<BR>; sip show registry Show
status of hosts we register with<BR>;<BR>; sip
debug Show all SIP messages<BR>;</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>[general]<BR>context=home ;
Default context for incoming calls</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>port=5060 ; UDP Port to bind to
(SIP standard port is 5060)<BR>bindaddr=0.0.0.0 ; IP address to bind
to (0.0.0.0 binds to all)<BR>srvlookup=yes ; Enable DNS SRV
lookups on outbound calls</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;[sip_proxy]<BR>; For incoming calls only. Example:
FWD (Free World Dialup)<BR>;type=user<BR>;context=from-fwd</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>;[sip_proxy-out]<BR>;type=peer
; we only want to call out, not be
called<BR>;secret=guessit<BR>;username=yourusername ; Authentication
user for outbound proxies<BR>;fromuser=yourusername ; Many SIP
providers require
this!<BR>;host=box.provider.com<BR>;------------------------------------------------<BR>;
Test Ext 2201</FONT></DIV>
<DIV><FONT face=Arial size=2>; <extension use> - <users name> -
<extension
number><BR>;------------------------------------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2> [2201]<BR> type=friend<BR> host=192.192.192.220<BR> context=home<BR> secret=xxxxxx<BR> callerid="Paul"
<2201><BR> mailbox=2201<BR> dtmfmode=rfc2833<BR> nat=no</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>EXTENSIONS.CONF</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>writeprotect=no</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[globals]<BR> PHONES1=SIP/2201<BR> PHONES1VM=2201<BR> PHONES2=SIP/2202<BR> PHONES2VM=2202<BR>CONSOLE=Console/dsp ;
Console interface for
demo<BR>;CONSOLE=Zap/1<BR>;CONSOLE=Phone/phone0<BR>IAXINFO=guest ;
IAXtel
username/password<BR>;IAXINFO=myuser:mypass<BR>TRUNK=Zap/g2 ;
Trunk interface<BR>TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)<BR></FONT><FONT face=Arial size=2></FONT></DIV>
<DIV><FONT face=Arial size=2>[iaxtel700]<BR>exten => _91700XXXXXXX,1,Dial(<A
href="mailto:IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel">IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel</A>)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[iaxprovider]<BR>;switch =>
IAX2/user:[key]@myserver/mycontext</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>[international]<BR><BR>; Master context for
international long distance<BR><BR>ignorepat => 9<BR>include =>
longdistance<BR>include => trunkint</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[longdistance]<BR><BR>; Master context for long
distance<BR><BR>ignorepat => 9<BR>include => local<BR>include =>
trunkld</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[local]<BR><BR>; Master context for local,
toll-free, and iaxtel calls only<BR>;<BR>ignorepat => 9<BR>include =>
default<BR>include => parkedcalls<BR>include => trunklocal<BR>include
=> iaxtel700<BR>include => trunktollfree<BR>include =>
iaxprovider<BR></FONT><FONT face=Arial size=2></FONT></DIV>
<DIV><FONT face=Arial size=2>;This will create a macro we will use in the
dialling plan<BR> [macro-vmessage]<BR> exten =>
s,1,VoiceMail2(u${ARG1})<BR> exten =>
s,2,Playback(groovy)<BR> exten => s,3,Playback(goodbye)<BR> exten
=> s,4,Hangup</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>[macro-stdexten];<BR>;<BR>; Standard extension
macro:<BR>; ${ARG1} - Extension (we could have used
${MACRO_EXTEN} here as well<BR>; ${ARG2} - Device(s) to
ring<BR>;<BR>exten => s,1,Dial(${ARG2},20) ;
Ring the interface, 20 seconds maximum<BR>exten =>
s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten =>
s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail
w/ unavail announce<BR>exten =>
s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to
start</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten =>
s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce<BR>exten => s-BUSY,2,Goto(default,s,1) ;
If they press #, return to start</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten =>
_s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten =>
a,1,VoicemailMain(${ARG1}) ; If they press *, send the
user into VoicemailMain</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;
----------------------------------------------<BR>; DEFINE EXTENSIONS<BR>;
----------------------------------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2> [home]<BR> ; Next, add an extension for
voicemail .<BR> ; now if we dial 8, we can check
voicemail.<BR> ;<BR> exten => 8,1,VoiceMailMain2<BR> exten
=> 8,2,Hangup<BR> ; Add some more extensions for the two lines . now
we'll be able to call one line from the other.<BR> ; And if no one answers,
it will go to the mailbox for that line.<BR> ;<BR> ; Line
1<BR> ;<BR> exten => 2201,1,Dial(${PHONES1},20,Ttm)<BR> exten
=> 2201,2,Macro(vmessage,${PHONES1VM})<BR> exten =>
2201,3,Hangup<BR> ;<BR> ; Line 2<BR> ;<BR> exten =>
2202,1,Dial(${PHONES2},20,Ttm)<BR> exten =>
2202,2,Macro(vmessage,${PHONES2VM})<BR> exten =>
2202,3,Hangup<BR> ;<BR> ; Line 3<BR> ;<BR> exten =>
2203,1,Dial(${PHONES3},20,Ttm)<BR> exten =>
2203,2,Macro(vmessage,${PHONES3VM})<BR> exten =>
2203,3,Hangup</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;
----------------------------------------------<BR>; END DEFINE EXTENSIONS<BR>;
----------------------------------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2><BR>[demo]<BR>;<BR>; We start with what to do when
a call first comes in.<BR>;<BR>exten => s,1,Wait,1 ; Wait a
second, just for fun<BR>exten => s,2,Answer ; Answer the
line<BR>exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5
seconds<BR>exten => s,4,ResponseTimeout,10 ; Set Response Timeout
to 10 seconds<BR>exten => s,5,BackGround(demo-congrats) ; Play a
congratulatory message<BR>exten => s,6,BackGround(demo-instruct) ; Play
some instructions</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten => 2,1,BackGround(demo-moreinfo) ;
Give some more information.<BR>exten => 2,2,Goto(s,6)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten => 3,1,SetLanguage(fr) ; Set
language to french<BR>exten => 3,2,Goto(s,5) ; Start with
the congratulations</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten => 1000,1,Goto(default,s,1)<BR>;<BR>; We
also create an example user, 1234, who is on the console and has<BR>; voicemail,
etc.<BR>;<BR>exten => 1234,1,Playback(transfer,skip) ; "Please
hold while..." <BR> ; (but skip if channel is not
up)<BR>exten => 1234,2,Macro(stdexten,1234,${CONSOLE})</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten => 1235,1,Voicemail(u1234) ;
Right to voicemail</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>exten => 1236,1,Dial(Console/dsp) ;
Ring forever<BR>exten => 1236,2,Voicemail(u1234) ; Unless
busy</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;<BR>; # for when they're done with the
demo<BR>;<BR>exten => #,1,Playback(demo-thanks) ; "Thanks for
trying the demo"<BR>exten => #,2,Hangup ; Hang them
up.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;<BR>; A timeout and "invalid extension
rule"<BR>;<BR>exten => t,1,Goto(#,1) ; If they take too
long, give up<BR>exten => i,1,Playback(invalid) ; "That's not
valid, try again"</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;<BR>; Create an extension, 500, for dialing
the<BR>; Asterisk demo.<BR>;<BR>exten => 500,1,Playback(demo-abouttotry); Let
them know what's going on<BR>exten => 500,2,Dial(<A
href="mailto:IAX2/guest@misery.digium.com/s@default">IAX2/guest@misery.digium.com/s@default</A>) ;
Call the Asterisk demo<BR>exten => 500,3,Playback(demo-nogo) ; Couldn't
connect to the demo site<BR>exten => 500,4,Goto(s,6) ; Return to
the start over message.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;<BR>; Create an extension, 600, for evaulating
echo latency.<BR>;<BR>exten => 600,1,Playback(demo-echotest) ; Let them
know what's going on<BR>exten => 600,2,Echo ; Do the echo
test<BR>exten => 600,3,Playback(demo-echodone) ; Let them know it's
over<BR>exten => 600,4,Goto(s,6) ; Start over</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;<BR>; Give voicemail at extension
8500<BR>;<BR>exten => 8500,1,VoicemailMain<BR>exten =>
8500,2,Goto(s,6)<BR>;<BR>; Here's what a phone entry would look like (IXJ for
example)<BR>;<BR>;exten => 1265,1,Dial(Phone/phone0,15)<BR>;exten =>
1265,2,Goto(s,5)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;[mainmenu]<BR>;<BR>; Example "main menu" context
with submenu<BR>;<BR>;exten => s,1,Answer<BR>;exten =>
s,2,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for
support, ..."<BR>;exten => 1,1,Goto(submenu,s,1)<BR>;exten =>
2,1,Hangup<BR>;include => default<BR>;<BR>;[submenu]<BR>;exten =>
s,1,Ringing ; Make them comfortable with 2 seconds
of ringback<BR>;exten => s,2,Wait,2<BR>;exten =>
s,3,Background(submenuopts) ; "Thanks for calling the sales
department. Press 1 for steve, 2 for..."<BR>;exten =>
1,1,Goto(default,steve,1)<BR>;exten => 2,1,Goto(default,mark,2)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>[default]<BR>;<BR>; By default we include the
demo. In a production system, you <BR>; probably don't want to have the
demo there.<BR>;<BR>include => demo</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;<BR>; Extensions like the two below can be used
for FWD, Nikotel, sipgate etc.<BR>; Note that you must have a [sipprovider]
section in sip.conf whereas<BR>; the otherprovider.net example does not require
such a peer definition<BR>;<BR>;exten =>
_41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)<BR>;exten =>
_42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>; Real extensions would go here. Generally you want
real extensions to be 4 or 5<BR>; digits long (although there is no such
requirement) and start with a single<BR>; digit that is fairly large (like 6 or
7) so that you have plenty of room to<BR>; overlap extensions and menu options
without conflict. You can alias them with<BR>; names, too and use global
variables</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;exten =>
6245,hint,SIP/Grandstream1&SIP/Xlite1 ; Channel hints for presence<BR>;exten
=> 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer<BR>;exten =>
6245,1,Dial(${HINT},20,rtT) ; Use hint as listed<BR>;exten =>
6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit<BR>;exten
=> 6389,1,Dial(<A
href="mailto:MGCP/aaln/1@192.168.0.14">MGCP/aaln/1@192.168.0.14</A>)<BR>;exten
=> 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>;exten =>
6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like
Zap/2<BR>;exten => mark,1,Goto(6275|1) ; alias mark to
6275<BR>;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for
wil<BR>;exten => wil,1,Goto(6236|1)<BR>;<BR>; Some other handy things are an
extension for checking voicemail via<BR>; voicemailmain<BR>;<BR>;exten =>
8500,1,VoicemailMain<BR>;exten => 8500,2,Hangup<BR>;<BR>; Or a conference
room (you'll need to edit meetme.conf to enable this room)<BR>;<BR>;exten =>
8600,1,Meetme(1234)<BR>;<BR>; Or playing an announcement to the called party, as
soon it answers<BR>;<BR>;exten =
8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))<BR>;<BR>; For more
information on applications, just type "show applications" at your<BR>; friendly
Asterisk CLI prompt.<BR>;<BR>; 'show application <command>' will show
details of how you<BR>; use that particular application in this file, the dial
plan. <BR>;</DIV>
<DIV><BR> <BR></DIV></FONT></BODY></HTML>