Hello,<br><br>In your sip.conf you don't have the user for you provider:<br><br>[yourprovider]<br>username=1234<br>secret=sdfdsf<br>host=<a href="http://sip.yourprovider.com">sip.yourprovider.com</a><br>type=peer<br>...........
<br><br>In yor extensions.conf<br><br>[mycontext]<br><br>exten => 2000,1,Dial(SIP/2000,20)<br>exten => 2000,103,Hangup<br><br>exten => 2001,1,Dial(SIP/2001,20)<br>exten => 2001,2,Hangup<br><br>exten => _001X.,1,Dial(
SIP/${EXTEN}@yourprovider,25)<br>exten => _001X.,2,Hungup()<br><br>For call out US, for example<br><br>Best Regards<br><div><span class="gmail_quote">On 6/18/07, <b class="gmail_sendername"><a href="mailto:achim@itsistem.ro">
achim@itsistem.ro</a></b> <<a href="mailto:achim@itsistem.ro">achim@itsistem.ro</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Dear asterisk users,<br><br>I need some help , I'm a little new in VoIP , asterisk. I have<br>downloaded, compiled , installed. I make a simple configuration (I'm sorry<br>write the configuration)<br>1. sip.conf<br>[general]
<br>port = 5060<br>bindaddr = <a href="http://0.0.0.0">0.0.0.0</a><br>allow=all<br>context=default<br>register => <a href="http://user:password@sip.myvoipprovider.com/2000">user:password@sip.myvoipprovider.com/2000</a>
<br><br>[2000]<br>type=friend<br>username=2000<br>secret=secret2000<br>host=dynamic<br>context=mycontex<br>maibox=2001<br><br>[2001]<br>type=friend<br>username=2001<br>secret=secret2001<br>host=dynamic<br>context=mycontext
<br>maibox=2001<br><br>2.extensions.conf<br>[general]<br>statis=yes<br>writeprotect=yes<br><br>[default]<br>exten => _.,1,Congestion<br><br>[mycontext]<br><br>exten => 2000,1,Dial(SIP/2000,20)<br>exten => 2000,103,Hangup
<br><br>exten => 2001,1,Dial(SIP/2001,20)<br>exten => 2001,2,Hangup<br><br>So far so good.Well the asterisk it's working , 2000 can call 2001 and<br>2001 cam call 2000 usind a VoiP ATA adapter or a softphone.Well my
<br>question is: I WANT TO WRITE A DIAL PLAN FOR USERS 2000 AND 2001 TO CALL A<br>NATIONAL NUMBER BY MY SIP PROVIDER <a href="http://sip.myvoipprovider.com">sip.myvoipprovider.com</a> (what do I have<br>to write in extensions.conf
anf how)<br>It's good a combination , like by pressing 0 , both users have a dial tone<br>for outside, if is to much<br><br>Thanks a lot<br><br>eng. Alexandru Achim<br>National Institute Lasers Physics<br>Quantum Solid States Laboratory
<br>Magurele,Bucharest,Romania<br><br><br>Sincerely<br><br><br><br><br>*****************************************************<br>Acest email a fost verificat de catre NOD32 Antivirus<br><br>Serviciu oferit de catre ITSISTEM SERVICES SRL
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