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

[Matt Watson]

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 RoLaNd
Sent: Wednesday, May 21, 2008 9:01 AM
To: asterisk-users at lists.digium.com
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 [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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