Michael Graves wrote:
On Thu, 30 Dec 2004 09:04:32 -0800, Steven P. Donegan wrote:
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system - however they don't get routed properly. The Asterisk message:
Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed
to authenticate user WIRELESS CALLER
<sip:[EMAIL PROTECTED]>;tag=7f8072c0c46250f7o1
X's are there to not advertise my phone number :-)
Any idea as to why any kind of authenticate would be done or would
fail would be appreciated.
Steven,
It really seems like you need to setup an entry in sip.conf that
"PSTN Line" on the sipura can register with. Do you have an entry in
sip.conf for it? How is "PSTN Line" programmed?
--
Kristian Kielhofner
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Here is sip show peers:
www*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port
Status
1004/1004 1.0.24.223 D 255.255.255.255 5060
Unmonitored
1003/1003 1.0.24.223 D 255.255.255.255 5060
Unmonitored
1002/1002 1.0.24.222 D 255.255.255.255 5061
Unmonitored
1001/1001 1.0.24.222 D 255.255.255.255 5060
Unmonitored
1000/1000 (Unspecified) D 255.255.255.255 0
Unmonitored
5 sip peers loaded [4 online , 1 offline]
Which seems to say the Sipura is registered...
Here is sip.conf:
[EMAIL PROTECTED] asterisk]# cat sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[1000]
type=friend
username=1000
fromuser=1000
host=dynamic
nat=no
canreinvite=yes
dtmfmode=rfc2833
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[1001]
type=friend
username=1001
fromuser=1001
host=dynamic
nat=no
canreinvite=yes
dtmfmode=rfc2833
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[1002]
type=friend
username=1002
fromuser=1002
host=dynamic
nat=no
canreinvite=yes
dtmfmode=rfc2833
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[1003]
type=friend
username=1003
secret=1003
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=1003
nat=no
disallow=all
allow=ulaw
[1004]
type=friend
username=1004
secret=1004
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=1004
nat=no
disallow=all
allow=ulaw
[EMAIL PROTECTED] asterisk]#
Not sure what I'm doing wrong but any suggestions would be welcomed.
And BTW - Happy Hollidays!
When I used the SPA-3000 I had to setup a special context in
extensions.conf and then use a "hotline" dialplan setup in the SPA.
This caused all calls incomming on the POTS line to immediately be
forwarded to the Asterisk context. I essentially bypassed the SPA
diaplan logic. You can find out more about this at www.voxilla.com
which hosts a forum for SPA users.
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
o713-861-4005
o800-905-6412
c713-201-1262
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Sorry for all the included text - but it is relevant. The problem is not
the Sipura->Asterisk connection - that is definitely happening - the
problem is that Asterisk seems to want to authenticate the call in some
way. And I have no clue at present as to how to make Asterisk happy
with the inbound call.
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
