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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Does your extensions.conf have any more configuration than what
you’ve shown?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>If not, then you are lacking dialplan for anything but internal
calls.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>--<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Matt<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>RoLaNd
RoLaNd<br>
<b>Sent:</b> Wednesday, May 21, 2008 9:01 AM<br>
<b>To:</b> asterisk-users@lists.digium.com<br>
<b>Subject:</b> [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to
pstn calls)<o:p></o:p></span></p>
</div>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>Hello all,<br>
<br>
its been a while im trying to setup my asterisk/sipura 3102 to recieve/make
calls from softphones on pcs in my home..<br>
i've set up 5 SIP extensions in sip.conf and made the dialing plan in
extensions.conf..<br>
i could make calls from 1 sip phone to another in my home.. but i cant call out
using pstn line interface nor recieve calls..<br>
please find below my topology as well as config info:<br>
<br>
(192.168.0.0)<br>
____________LAN______________<br>
|
|
|<br>
softphone
asterisk
sipura---------PSTN LINE<br>
<br>
<br>
<br>
Configuration:<br>
<br>
ASTERISK: <br>
<br>
sip.conf <br>
<br>
[101] <br>
type=peer <br>
port=5062 <br>
host=dynamic <br>
secret=1234 <br>
context=spa <br>
<br>
<br>
[103] <br>
type=peer <br>
port=5061 <br>
host=dynamic <br>
secret=1234 <br>
context=spa <br>
<br>
[100] <br>
type=peer <br>
port=5061 <br>
host=dynamic <br>
secret=1234 <br>
context=spa <br>
<br>
[111] <br>
type=peer <br>
port=5060 <br>
host=dynamic <br>
secret=1234 <br>
context=spa <br>
<br>
================================================== =========== <br>
<br>
EXTENSIONS.CONF <br>
<br>
[spa] <br>
Exten => _1XX,1,Dial(SIP/${EXTEN}) <br>
<br>
================================================== =========== <br>
<br>
<br>
and this is the settings i have right now for sipura 3102 in my PSTN LINE: <br>
<br>
<br>
<a
href="http://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg"
target="_blank">http://img84.imageshack.us/my.php?image=40541922um2.jpg</a> <br>
<br>
<a
href="http://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg"
target="_blank">http://img98.imageshack.us/my.php?image=55448347ss9.jpg</a> <br>
<br>
<a href="http://img262.imageshack.us/my.php?imag%20...%20472qz3.jpg"
target="_blank">http://img262.imageshack.us/my.php?imag ... 472qz3.jpg</a> <br>
<br>
ps: i read so many tutorials and none seems to help..<br>
lately whenever i try to call out using my sipphone.. it gives me "503
service unavailable" and this is wht shows on the CLI of asterisk when i
set sip debug on..<br>
<br>
<br>
<br>
<br>
ubuntu-pbx-desktop*CLI><br>
== Connect attempt from '127.0.0.1' unable to authenticate<br>
-- Executing [1009@spa:1] Dial("SIP/1003-b5f05600",
"SIP/1009") in new stack<br>
-- Called 1009*CLI><br>
-- Got SIP response 410 "Gone" back from
192.168.0.111<br>
-- SIP/1009-081741d0 is circuit-busy<br>
== Everyone is busy/congested at this time (1:0/1/0)<br>
== Auto fallthrough, channel 'SIP/1003-b5f05600' status is 'CONGESTION'<br>
<br>
<o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Invite
your mail contacts to join your friends list with Windows Live Spaces. It's
easy! <a
href="http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us"
target="_new">Try it!</a><o:p></o:p></span></p>
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