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<body class='hmmessage'>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 onclick="urchinTracker ('/outgoing/http_www_voipuser_org_ship_to_php_url_http_img84_imageshack_us_my_php_image_40541922um2_jpg');" href="http://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg" target=_blank><U><FONT face="" color=#0000ff>http://img84.imageshack.us/my.php?image=40541922um2.jpg</FONT></U></A> <BR><BR><A onclick="urchinTracker ('/outgoing/http_www_voipuser_org_ship_to_php_url_http_img98_imageshack_us_my_php_image_55448347ss9_jpg');" href="http://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg" target=_blank><U><FONT face="" color=#0000ff>http://img98.imageshack.us/my.php?image=55448347ss9.jpg</FONT></U></A> <BR><BR><A onclick="urchinTracker ('/outgoing/http_www_voipuser_org_ship_to_php_url_http_img262_imageshack_us_my_php_image_43929472qz3_jpg');" href="http://img262.imageshack.us/my.php?imag ... 472qz3.jpg" target=_blank><U><FONT face="" color=#0000ff>http://img262.imageshack.us/my.php?imag ... 472qz3.jpg</FONT></U></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>
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