Given that he is using plaintext as the auth method, I guess anyone
who wants that
password can have it by snooping anyhow. :-)
T.
On 1 Jun 2009, at 07:18, Rob Hillis wrote:
The clue in the log is "no authority found". Something in the
configuration at the other end doesn't match the config
The clue in the log is "no authority found". Something in the
configuration at the other end doesn't match the configuration at this
end - almost certainly the username and password.
Why are you including the IP address when dialling the trunk? If your
peers are set up with IP addresses (which t
my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go
to 1.6 voice mail. it says
== Using SIP RTP CoS mark 5
-- Executing [4...@sip:1] Dial("SIP/312-09f9a720",
"IAX2/trun...@147.120.203.98/4567,10,t") in new stack
-- Called trun...@147.120.203.98/4567
[Jun 1 11
Asterisk versions may differ. I do IAX trunk successfully even
between Asterisk 1.0.2 and 1.4.xx
please post your Dial command.
On Fri, May 29, 2009 at 11:33 AM, Tharanga wrote:
> Hi All,
>
> Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and
> asterisk 1.2.14 ?
>
> i tried
Hi All,
Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and
asterisk 1.2.14 ?
i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but
it gave an error -
1.2.14 End - Error Msg
WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by
147.120.203.71: No
