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<body class='hmmessage'>yes thats the only thing i have in extensions.conf<BR>
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should there be anything else?! <BR>
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Message: 21<BR>Date: Wed, 21 May 2008 09:40:26 -0400<BR>From: Matt Watson <<A href="mailto:mwatson@becon.org">mwatson@becon.org</A>><BR>Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to<BR> sip/sip to pstn calls)<BR>To: Asterisk Users Mailing List - Non-Commercial Discussion<BR> <<A href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>><BR>Message-ID:<BR> <<A href="mailto:60BDBA04C2769C49AD1416789D9123A00F9884C9F8@columbia.becon.int">60BDBA04C2769C49AD1416789D9123A00F9884C9F8@columbia.becon.int</A>><BR>Content-Type: text/plain; charset="us-ascii"<BR> <BR>Does your extensions.conf have any more configuration than what you've shown?<BR> <BR>If not, then you are lacking dialplan for anything but internal calls.<BR> <BR>--<BR>Matt<BR> <BR>From: <A href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A> [mailto:<A href="mailto:asterisk-users-bounces@lists.digium.com]">asterisk-users-bounces@lists.digium.com]</A> On Behalf Of RoLaNd RoLaNd<BR>Sent: Wednesday, May 21, 2008 9:01 AM<BR>To: <A href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A><BR>Subject: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)<BR> <BR>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=onClickUnsafeLink(event); href="http://img84.imageshack.us/my.php?image=40541922um2.jpg" target=_blank>http://img84.imageshack.us/my.php?image=40541922um2.jpg</A><<A onclick=onClickUnsafeLink(event); href="http://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg" target=_blank>http://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg</A>><BR> <BR><A onclick=onClickUnsafeLink(event); href="http://img98.imageshack.us/my.php?image=55448347ss9.jpg" target=_blank>http://img98.imageshack.us/my.php?image=55448347ss9.jpg</A><<A onclick=onClickUnsafeLink(event); href="http://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg" target=_blank>http://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg</A>><BR> <BR><A onclick=onClickUnsafeLink(event); href="http://img262.imageshack.us/my.php?imag" target=_blank>http://img262.imageshack.us/my.php?imag</A> ... 472qz3.jpg<<A onclick=onClickUnsafeLink(event); href="http://img262.imageshack.us/my.php?imag ... 472qz3.jpg" target=_blank>http://img262.imageshack.us/my.php?imag%20...%20472qz3.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 /><hr />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></body>
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