[asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)

[RoLaNd RoLaNd]

yes thats the only thing i have in extensions.conf
 
should there be anything else?! 
 
 
Message: 21Date: Wed, 21 May 2008 09:40:26 -0400From: Matt Watson <mwatson at becon.org>Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to	sip/sip to pstn	calls)To: Asterisk Users Mailing List - Non-Commercial Discussion	<asterisk-users at lists.digium.com>Message-ID:	<60BDBA04C2769C49AD1416789D9123A00F9884C9F8 at columbia.becon.int>Content-Type: text/plain; charset="us-ascii" 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of RoLaNd RoLaNdSent: Wednesday, May 21, 2008 9:01 AMTo: asterisk-users at lists.digium.comSubject: [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=peerport=5062host=dynamicsecret=1234context=spa  [103]type=peerport=5061host=dynamicsecret=1234context=spa [100]type=peerport=5061host=dynamicsecret=1234context=spa [111]type=peerport=5060host=dynamicsecret=1234context=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 [1009 at spa: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'
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