Hi John, <br><br>Your first question, I am not sure why ....but for this part i can explain abit <br><blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;" class="gmail_quote">
Also, on a side note, I have a context called [home] which each SIP<br>Phone is associated with. Do I need to specify each extension in<br>there?<br></blockquote><br>SIP user can register as name as well . Doesnt means to have number. Example in
sip.conf <br>[john]<br>type=friend<br>username=john<br>host=dynamic<br>context=incoming<br>secret=6769<br>dtmfmode=rfc2833<br>disallow=all<br>allow=ulaw<br>insecure=very<br><br>Then in extensions.conf , you can have any number to ring this john sip user phone. Example :
<br><br>exten =>9XXXXXXX,1,Dial(SIP/john) ; any number start with 9 end with 7 digit behinds. or you can also <br>exten => 9XXXXXXX,2,Hangup<br><br>exten => s,1,Dial(SIP/john) ; starting of the incoming call will ring John phone.
<br>exten => s,2,Hangup<br><br>Hope my explaination is clear or fullfill your needs....thanks<br><br><div><span class="gmail_quote">On 6/17/06, <b class="gmail_sendername">John Klimek</b> <<a href="mailto:jklimek@gmail.com">
jklimek@gmail.com</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;">Incoming calls from my Sipura 3000 don't seem to be correctly routing
<br>to Asterisk (or something?)<br><br>Here is my Asterisk configuration for my incoming PSTN line:<br>Code:<br><br>[1000]<br>type=friend<br>host=dynamic<br>context=incoming<br>secret=6769<br>dtmfmode=rfc2833<br>disallow=all
<br>allow=ulaw<br>insecure=very<br><br><br>Inside of extensions.conf, I have this:<br>Code:<br><br>[incoming]<br>exten => s,1,Answer( )<br>exten => s,2,Background(enter-ext-of-person)<br><br><br>When I call my PSTN line, my Sipura 3000 seems to successfully answer
<br>it because the line rings once, but then immediately switches to a<br>second dial tone. Shouldn't my incoming call be answered and then have<br>"enter-ext-of-person" played to them?<br><br>What could be causing this?
<br><br>Also, on a side note, I have a context called [home] which each SIP<br>Phone is associated with. Do I need to specify each extension in<br>there?<br><br>For example:<br><br>exten => 50,1,Dial(SIP/50)<br>exten => 50,2,Hangup
<br><br>exten => 21,1,Dial(SIP/21)<br>exten => 21,2,Hangup<br><br>Can't I just setup a default system where any two-digit number is<br>assumed to be an extension and it is automatically tried?<br><br>Thanks for any help!!
<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:
<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br>