I was able to get on the UI of the Yealink T32G and fiddle with the
setting. Here's the setting for TLS transport in
/etc/asterisk/extensions.conf:
[transport-tls]
type = transport
protocol = tls
bind = 0.0.0.0:5061
; ca_list_file = /etc/asterisk/keys/ca.crt
; cert_file =
Thanks Joshua for the tip re using hostname rather than IP address when
configuring the phone. It worked nicely on the linphone on my macbookpro
at home. Dialplans are followed faithfully w/o the problems I experienced
earlier. I'll test using the hostname on the Yealink phone next time I'm
in
On Thu, Feb 11, 2021 at 9:01 PM Ruisheng Peng wrote:
> Sorry, my bad. I failed to change the transport to tls on the provision
> for the hardphone, nor did change the transport on the linphone setup.
> However, after I do that, the hardphone (Yealink T32G) failed to register,
> citing:
>
> [Feb
Sorry, my bad. I failed to change the transport to tls on the provision
for the hardphone, nor did change the transport on the linphone setup.
However, after I do that, the hardphone (Yealink T32G) failed to register,
citing:
[Feb 11 14:16:03] WARNING[24936]: pjproject: :SSL
On Mon, Feb 8, 2021 at 6:14 PM Ruisheng Peng wrote:
> Thanks Jashua for the suggestion. To find out if the issue was only
> limited to the softphone that was using tls transport (SOFTPHONE_B on ext
> 103, a linphone running off my MBP), I also turned one of the hard phone
> (f30A0A01 on ext
Thanks Jashua for the suggestion. To find out if the issue was only
limited to the softphone that was using tls transport (SOFTPHONE_B on ext
103, a linphone running off my MBP), I also turned one of the hard phone
(f30A0A01 on ext 100, a Yealink T32G) into using tls transport. It
behaves
On Wed, Feb 3, 2021 at 11:02 PM Ruisheng Peng wrote:
When using handsets with udp or tcp transports to dial ext 100, it'd hangup
> after the no-one-arround message. However, when using the handset with tls
> transport, it doesn't hang up on its own if ext 100 is not answered. I
> have to
Hi all,
I managed to get tls transport going with asterisk 16.14.0, and set one
handset (SOFTPHONE_B) to use the transport. I have set up a few other
handsets (both soft and hard) to use udp and tcp transports:
voip1*CLI> pjsip show endpoints
Endpoint:
I/OAuth:
Aor:
