Does your extensions.conf have any more configuration than what you've shown?
If not, then you are lacking dialplan for anything but internal calls. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd Sent: Wednesday, May 21, 2008 9:01 AM To: [email protected] Subject: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls) Hello all, its been a while im trying to setup my asterisk/sipura 3102 to recieve/make calls from softphones on pcs in my home.. i've set up 5 SIP extensions in sip.conf and made the dialing plan in extensions.conf.. 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.. please find below my topology as well as config info: (192.168.0.0) ____________LAN______________ | | | softphone asterisk sipura---------PSTN LINE Configuration: ASTERISK: sip.conf [101] type=peer port=5062 host=dynamic secret=1234 context=spa [103] type=peer port=5061 host=dynamic secret=1234 context=spa [100] type=peer port=5061 host=dynamic secret=1234 context=spa [111] type=peer port=5060 host=dynamic secret=1234 context=spa ================================================== =========== EXTENSIONS.CONF [spa] Exten => _1XX,1,Dial(SIP/${EXTEN}) ================================================== =========== and this is the settings i have right now for sipura 3102 in my PSTN LINE: http://img84.imageshack.us/my.php?image=40541922um2.jpg<http://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg> http://img98.imageshack.us/my.php?image=55448347ss9.jpg<http://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg> http://img262.imageshack.us/my.php?imag ... 472qz3.jpg<http://img262.imageshack.us/my.php?imag%20...%20472qz3.jpg> ps: i read so many tutorials and none seems to help.. 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.. ubuntu-pbx-desktop*CLI> == Connect attempt from '127.0.0.1' unable to authenticate -- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f05600", "SIP/1009") in new stack -- Called 1009*CLI> -- Got SIP response 410 "Gone" back from 192.168.0.111 -- SIP/1009-081741d0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1003-b5f05600' status is 'CONGESTION' ________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it!<http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us>
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