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<DIV><FONT face=Arial size=2>The Asrerisk config which is tested and
working is given below. The system has</FONT></DIV>
<DIV>1). X100P - Card<BR>2). Two -Greadstream100 SIP Phones.</DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Asterisk config.</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>writeprotect=no</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>; You can include other config files, use the
#include command (without the ';')<BR>; Note that this is different from the
"include" command that includes contexts within<BR>; other contexts. The
#include command works in all asterisk configuration files.<BR>;#include
"filename.conf"<BR>; The "Globals" category contains global variables that can
be referenced<BR>; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for
Environmental variable<BR>; ${${VARIABLE}} or ${text${VARIABLE}} or any
hybrid<BR>;<BR>[globals]<BR>Trunk=Zap/1<BR>phone1=SIP/197<BR>phone2=SIP/198<BR>;phone3=SIP/131<BR>everyone=${phone1}&${phone2}<BR>[from-pstn]<BR>exten=>_0.,1,Dial($Trunk/$everyone,13,tTmr)<BR>exten=>_0.,2,Congestion<BR>;exten=>_0.,1,Dial($Trunk/phone2,13,tTmr)<BR>;exten=>_0.,2,Congestion<BR>;ignorepat
=> 0</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV> </DIV><FONT face=Arial size=2>
<DIV><BR>[from-sip]<BR>;[incoming]<BR>;exten =>
_0.,Dial(Zap/1/(${exten})<BR>;exten => _0.,2,Hangup<BR>;exten =>
197,1,Dial(SIP/197,20,tr)<BR>;exten => 197,2,Hangup<BR>;exten =>
198,1,Dial(SIP/198,20,tr)<BR>;exten => 198,2,Hangup<BR>;</DIV>
<DIV> </DIV>
<DIV>;exten=>_.,1,Dial(Zap/1/SIP/197,20,tT)<BR>;exten=>_.,1,Dial(Zap/1/SIP/198,20,tT)</DIV>
<DIV> </DIV>
<DIV>[incoming]<BR>;exten => 131,1,Dial(SIP/131,20,tr)<BR>;exten =>
131,2,Hangup<BR>exten => 197,1,Dial(SIP/197,20,tr)<BR>exten =>
197,2,Hangup<BR>exten => 198,1,Dial(SIP/198,20,tr)<BR>exten =>
198,2,Hangup</DIV>
<DIV> </DIV>
<DIV><BR>exten=>_xxxxxxxxxxx,1,Dial(${Trunk}/${EXTEN}),20,tT)<BR>exten=>_xxxxxxxxxxx,2,Hangup</DIV>
<DIV> </DIV>
<DIV>Sip.conf</DIV>
<DIV>;<BR>[general]<BR>auth=plaintext<BR>qualify=no<BR>nat=yes</DIV>
<DIV> </DIV>
<DIV>;phone 1 Grandstream
Phone<BR>[131]<BR>port=5060<BR>type=peer<BR>type=user<BR>;context=internalsipphones<BR>context=from-sip<BR>host=dynamic<BR>defaultip=212.135.237.131<BR>canreinvite=yes<BR>disallow=all<BR>allow=ulaw<BR>allow=alaw<BR>allow=gsm<BR>;<BR>;<BR>;phone
2 Grandstream
Phone<BR>[198]<BR>port=5060<BR>type=peer<BR>type=user<BR>;context=internalsipphones<BR>context=from-sip<BR>host=dynamic<BR>defaultip=217.37.237.198<BR>canreinvite=yes<BR>disllow=all<BR>allow=ulaw<BR>allow=alaw<BR>allow=gsm<BR>;<BR>Zapata.conf[channels]<BR>usecallerid=yes<BR>callerid=asreceived<BR>hidecallerid=no<BR>callwaiting=yes<BR>callwaitingcallerid=yes<BR>threewaycalling=yes<BR>transfer=yes<BR>cancallforward=yes<BR>callreturn=yes<BR>immediate=no<BR>callprofress=no<BR>echotraining=yes<BR>echocancel=yes<BR>echocancelwhenbridge=yes<BR>switchtype=national<BR>signalling=fxs_ks<BR>context=from-pstn<BR>cidsignalling=v23<BR>cidstart=history<BR>group=1<BR>musiconhold=default<BR>channel
=>1<BR>~<BR>~<BR>~<BR>~<BR>Zaptel.conf</DIV>
<DIV> </DIV>
<DIV>#<BR>loadzone = uk<BR>#loadzone =
us-old<BR>#loadzone=gr<BR>#loadzone=it<BR>#loadzone=fr<BR>#loadzone=de<BR>#loadzone=uk<BR>#loadzone=fi<BR>#loadzone=jp<BR>#loadzone=sp<BR>#loadzone=no<BR>defaultzone=uk<BR>fxsks
=1</DIV>
<DIV> </DIV>
<DIV>The system is for UK config.</DIV>
<DIV> </DIV>
<DIV>appan kh</DIV>
<DIV></FONT> </DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=ast_user0@yahoo.com href="mailto:ast_user0@yahoo.com">Nil S</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, September 23, 2005 9:02
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] SIP exten
to PSTN calls</DIV>
<DIV><BR></DIV>
<DIV>
<DIV>Hello,</DIV>
<DIV> </DIV>
<DIV>I have read your email.</DIV>
<DIV> </DIV>
<DIV>I found that you have configured X100P card and established a call from
SIP exten. to SIP exten and PSTN to SIP exten.</DIV>
<DIV> </DIV>
<DIV>I have done the first part i.e. SIP exten to SIP exten and would like to
do a second part. So please help me regarding this.</DIV>
<DIV> </DIV>
<DIV>I have installed Asterisk on Linux machine. So from here please guide me
how i should proceed. What are the requirements? and some other details.</DIV>
<DIV> </DIV>
<DIV>Your help will be much appriciated.</DIV>
<DIV> </DIV>
<DIV>Thanks,</DIV>
<DIV>Nil.<BR><BR><B><I>Appan KH <appan@softswitches.net></I></B>
wrote:</DIV>
<BLOCKQUOTE class=replbq
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Hi,<BR>I
had configured Asterisk with the following<BR>1). X100P - Card<BR>2). Two
-Greadstream100 SIP Phones.<BR>I am able to make calls from SIP Ext to SIP
Ext and PSTN calls from outside <BR>to SIP Extn.<BR>But I am not able to
make calls from SIP Extn to PSTN out going calls-it <BR>gives BT error
message- The number you had dialled not recognised.<BR>The SIP extn is not
sending the correct number.<BR>I will be thank full if some solutions is
suggested.<BR><BR>appan
kh<BR><BR><BR>_______________________________________________<BR>Asterisk-Users
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list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To
UNSUBSCRIBE or update options
visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE></DIV>
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