[asterisk-users] Peer is UNREACHABLE
kevin.larsen at pioneerballoon.com
Thu May 28 16:02:17 CDT 2015
> No, I'm not sure.
> And no, I can't make any call, right now... At least, not connected to
> If I connect it to the other VM with AsteriskNOW I can call my Twinkle,
> NOT my phone connected on my Asterisk, using the "proxy".
> I can see that in the log:
> [May 28 22:49:51] WARNING: chan_sip.c:12800 check_auth: username
> mismatch, have <1234>, digest has <luca>
> [May 28 22:49:51] NOTICE: 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 parts that you need to have in place for the basics to work.
You need your sip.conf in order to tell asterisk what devices and phone
trunks you have and you need extensions.conf to tell Asterisk how to route
calls. Since you are new to this, you can start by getting the two phones
to both register (sounds like one of them is and one probably is not).
Then you get to where you can dial from one phone to the other and vice
versa. From there you can add in the telephone company lines and the
ability to dial in and out to the world.
I am still curious why you have both an Asterisk setup and an AsteriskNow
setup? Is that just to play around with? At the end of the day you should
just need one or the other.
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