I figured out my problem. Cleared all browser settings at one point. Once I
visited the secure IP address 8089/ws web page directly (accepting unsafe
browsing) I was able to login.
From: asterisk-users On Behalf Of Dan
Cropp
Sent: Wednesday, December 12, 2018 10:52 AM
To: asterisk-users@lis
On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote:
>
> The problem: The extension doesn't create a ringback locally, because
> it most probably expects it to
> be sent by the callee - but the callee doesn't send anything (not
> surprising, because there has been
> no SDP).
>
> Or should
Howdy,
Aye. I've put an example over here:
https://wiki.asterisk.org/wiki/display/DIGIUM/Provisioning
look for the expand under "Here is an example Avahi services definition
file..."
Cheers
On Wed, Dec 12, 2018 at 11:21 AM Mitch Claborn wrote:
> I'm working on an asterisk upgrade to 16.1 and
I'm working on an asterisk upgrade to 16.1 and am remote from that
location. We use Digium phones there, configured with DPMA. From my VPN
I can connect to the server directly with the phone on my desk, but it
doesn't find the configuration server automatically since I'm on a
different physical
I had SIPML5 working with my Asterisk 16 last week. Not sure what I changed,
but I'm now receiving the following in asterisk whenever I try to login.
Can anyone provide some guidance on what I should be looking at or how to
diagnose the problem?
[12/12 08:46:18.161] DEBUG[7322] http.c: HTTP op
Hello!
An extension registered at asterisk 13.23 initiates an external call (pjsip).
After the Invite, the
callee (-> ISP) sends a
100 Trying
183 Session Progress (*without* SDP)
Asterisk now sends to the extension:
183 Session Progress (*with* SDP)
183 Session