[asterisk-users] Failed to authenticate
Antony Stone
Antony.Stone at asterisk.open.source.it
Mon Aug 9 10:14:11 CDT 2021
On Monday 09 August 2021 at 17:05:42, Jerry Geis wrote:
> Well my issue has changed now. I have dropped the 3382. Changed back to
> 3381. So I am registering 3381 to the other server.
That makes more sense to me, at least.
> The other server is 10.35.229.5. My IP is 10.35.229.11.
> I have two network cards.
>
> 10.35.229.11 is Eth0
> 192.168.1.60 is Eth1
>
> route looks OK
I think eth1 and your routing table are not relevant to this.
> The issue is that the call comes in but the user hears no audio.
> There is any crazy networking going on - why would the user not hear audio?
Commonly, because of firewalling and/or NAT.
Given that your client 10.35.229.11/24 and the server 10.35.229.5/24 are both
on the same subnet, it's not going to be a NAT problem, so I would look at the
firewall rules, both on your machine and the one you are connecting to.
(Please do tell us if the client and server are not connected directly through
a switch as I have assumed here, and there's possibly something more
complicated going on.)
You want to look for firewall rules which will allow UDP in both directions on
ports 10000 - 30000 (typically, may vary a bit, but something like that), or
alternatively, look for any rules which would block this, and remove them.
If that doesn't appear to be the problem, do a packet capture of your SIP
traffic and look for the Invite and the reply, each with the SDP payloads, to
find out what IP addresses and port numbers the client and server are
advertising to each other.
The only other thing I can think of right now is codec compatibility.
Antony.
--
I know I always wanted to be somebody, but I guess I should have been more
specific.
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