SIP provider is not receiving Asterisk replies to Options Polling. The SIP is running TLS and is registered. However, calling DID’s do not reach asterisk server. SIP provider indicates Asterisk is not replying to option polling. Asterisk logs indicate Asterisk is replying to Option polling.
Reliably Transmitting (no NAT) to 162.223.83.245:5061:
OPTIONS sip:162.223.83.245 SIP/2.0
Via: SIP/2.0/TLS 52.232.129.229:5061;branch=z9hG4bK011ec68b
Max-Forwards: 70
From: “asterisk” sip:asterisk@52.232.129.229;tag=as558b8503
To: sip:162.223.83.245
Contact: sip:asterisk@52.232.129.229:5061;transport=tls
Call-ID: 42e633c83044ca1675913e2c01e7c81e@52.232.129.229:5061
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.1
Date: Wed, 20 Feb 2019 14:57:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from TLS:162.223.83.245:5061 —>
SIP/2.0 200 OK
From: “asterisk” sip:asterisk@52.232.129.229;tag=as558b8503
To: sip:162.223.83.245;tag=sip+2+b0080104+c35c3595
Via: SIP/2.0/TLS 52.232.129.229:5061;branch=z9hG4bK011ec68b
Server: SIP/2.0
Max-Forwards: 70
Contact: sip:asterisk@52.232.129.229:5061;transport=tls
Call-ID: 42e633c83044ca1675913e2c01e7c81e@52.232.129.229:5061
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.1
Date: Wed, 20 Feb 2019 14:57:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘42e633c83044ca1675913e2c01e7c81e@52.232.129.229:5061’ Method: OPTIONS
The SIP provider ask if Asterisk could be configured to reply to option polling by way of a different SIP proxy address over UDP. While still maintaining TLS SIP signaling and Voice.