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 <[EMAIL
PROTECTED]>Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to
sip/sip to pstn calls)To: Asterisk Users Mailing List - Non-Commercial
Discussion <[email protected]>Message-ID: <[EMAIL
PROTECTED]>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:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNdSent:
Wednesday, May 21, 2008 9:01 AMTo: [EMAIL PROTECTED]: [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 [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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