[asterisk-users] Queue don't call Interface PJSIP

Roberto roberto.medola at gasparimsantos.com.br
Mon Aug 17 16:15:38 CDT 2020


Hello.


I am having a lot of problems with SIP through NAT. So, I decided to 
adopt PJSIP. However, I am not able to make the extensions ring when 
receiving a call from the queue. I'm using telnet to include the 
extension and on the asterisk console, it even shows Called PJSIP/6001, 
but the extension doesn't ring. If I call from extension to extension, 
it works normally.

telenet:
Action: QueueAdd
Queue: queuetest
MemberName: 1234
Interface: PJSIP/6001
StateInterface: PJSIP/6001
Ringinuse: yes
Paused: false

If I change to SIP, the extension will call normally.

My configuration pjsip.conf

[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0:5160
local_net=192.0.0.0/24
external_media_address=192.168.0.196
external_signaling_address=192.168.0.196

[6001]
type=endpoint
context=callcenter
disallow=all
allow=g729
allow=ulaw
allow=gsm
auth=6001
aors=6001
transport=transport-udp-nat
direct_media=no
allow_subscribe=yes
sub_min_expiry=30

[6001]
type=auth
auth_type=userpass
password=6001
username=6001

[6001]
type=aor
max_contacts=99


Has anyone experienced the same problem? I upgraded my asterisk to 17 
and the problem still persists.

Thanks!

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