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://img98.imageshack.us/my.php?image=55448347ss9.jpg
http://img262.imageshack.us/my.php?imag ... 472qz3.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'
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