TLS/SRTP, getting busy signal

Asterisk Asterisk Support
1

I’ve enabled SRTP in my Asterisk 13 based server.
Detail specification is here:

CentOS: 7 64bit
FreePBX: 13.0.97	
Asterisk: 13.7.2

pjsip.transport.conf

[0.0.0.0-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
external_media_address=52.*.*.*
external_signaling_address=52.*.*.*
ca_list_file=/etc/asterisk/keys/integration/ca-bundle.crt
cert_file=/etc/asterisk/keys/integration/webserver.crt
priv_key_file=/etc/asterisk/keys/integration/webserver.key
local_net=172.31.30.109/255.255.240.0
local_net=172.31.30.109
method=tlsv1

Under Extension > 12345 -> Advanced -> Media Encryption -> SRTP via in SDP

SIP client is being registered successfully using TLS transport type & Encryption Enabled. RPRT for signaling and media was enabled.

I’m getting busy signal when trying to make call. And not a single output in asterisk console with debug mode on.

I tried same thing in several servers, all are working fine. Not sure if any changes made in recent version of asterisk or freepbx.

I need your help.

Thanks

2

If you do “pjsip set logger on” do you see any SIP traffic when attempting to place a call? If not then it seems as though the device isn’t even sending the call to Asterisk…

3

Not sure what happened but I’ve reinstalled and same settings are working fine now.