Hello Friends,
I am using FreePBX 14.0.13.6. I am not getting any ringback tone when I call any of my extensions from outside.
My ISP is sending 180 response and they also said that the issue will in the server.
Thank in advance.
This is my incoming info
pbx3*CLI> sip set debug peer
<— Transmitting (NAT) to 10.xxx.xxx.xxx:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.xxx.xxx.xxx:5060;branch=z9hG4bKdu1s4bous4t21t3r44b9wsbmr;Role=3;Hpt=8eb2_36;TRC=ffffffff-ffffffff;received=10.xxx.xxx.xxx;rport=5060
From: sip:[email protected];tag=obnyzywc
To: sip:[email protected];tag=as1056712d
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-14.0.13.6(13.27.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
– Connected line update to SIP/Airtel-in-00000006 prevented.
– SIP/4036900-00000007 is ringing
<— Transmitting (NAT) to 10.xxx.xxx.xxx:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.xxx.xxx.xxx:5060;branch=z9hG4bKdu1s4bous4t21t3r44b9wsbmr;Role=3;Hpt=8eb2_36;TRC=ffffffff-ffffffff;received=10.xxx.xxx.xxx:5060;rport=5060
From: sip:[email protected];tag=obnyzywc
To: sip:[email protected];tag=as1056712d
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-14.0.13.6(13.27.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0