Displaying 4 results from an estimated 4 matches for "as6dd12e05".
2015 May 28
4
Peer is UNREACHABLE
...that in the log:
[May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username
mismatch, have <1234>, digest has <luca>
[May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite:
Failed to authenticate device "Test1" <sip:1234 at 172.16.34.132>;tag=as6dd12e05
Thanks
Luca Bertoncello
(lucabert at lucabert.de)
2015 May 28
0
Peer is UNREACHABLE
...t; [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username
> mismatch, have <1234>, digest has <luca>
> [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite:
> Failed to authenticate device "Test1" <sip:1234 at 172.16.34.132>;tag=as6dd12e05
>
> Thanks
> Luca Bertoncello
> (lucabert at lucabert.de)
>
2015 May 28
0
Peer is UNREACHABLE
...; [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username
> mismatch, have <1234>, digest has <luca>
> [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite:
> Failed to authenticate device "Test1"
<sip:1234 at 172.16.34.132>;tag=as6dd12e05
>
I know from your previous email that you are new to Asterisk. Have you
created a dialplan that would allow you to call from one extension to
another without going through your phone company? That is to say, call
from your phone through Asterisk to your wife's phone?
You have two part...
2015 May 28
4
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb:
> I'd start by turning on sip debugging in asterisk
> >sip set debug ip [your_phone_ip]
Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.200.11:5060:
OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0
Via: SIP/2.0/UDP