Incoming calls do not have any ACK messages

AtronSeige · October 10, 2019, 10:16am

We have an issue where an incoming call is “registered” on Asterisk, but when the server returns the invite, there is never an ACK. The call is Answered on the server side (with monkeys), but the caller’s phone just dies.

We can make outgoing calls and all is fine.

Asterisk 16.5.1
Ubuntu Server 18
No firewalls are enabled on the machine, firewall is a separate server.

Below is a capture of the logs for the incoming call. The IPs and such have been removed.

Full:
[Oct 10 11:41:19] VERBOSE[24646] res_pjsip_logger.c: <— Received SIP request (1322 bytes) from UDP:[InternalRouterIP]:5060 —>
INVITE sip:[SipUser]@[ServerIP]:5060 SIP/2.0
Via: SIP/2.0/UDP [SipProviderOut]:5060;branch=z9hG4bK-524287-1—18dc27db2247d16c477e43d41006afbe;rport
Via: SIP/2.0/UDP [SipProviderIn]:5070;branch=z9hG4bK-5hr4zfmtdhf3reig;rport=5070
Max-Forwards: 69
Record-Route: sip:SipProviderOut;lr;ep
Contact: sip:[SipProviderIn]:5070
To: sip:[SipUser]@SipProvider
From: sip:[CallingNumber]@SipProvider;tag=R2b8Jl6YKAndTzPk.o
Call-ID: 0a57cd0f16@41.87.218.2~1o~3o
CSeq: 914 INVITE
Expires: 300
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Disposition: session
Content-Type: application/sdp
User-Agent: PortaSIP
P-Asserted-Identity: sip:[CallingNumber]@SipProvider
Remote-Party-ID: sip:[CallingNumber]@SipProvider;party=calling
h323-conf-id: 3553874358-2583339329-2779392248-2779392248
PortaSIP-Notify: aor=[Username]
cisco-GUID: 3553874358-2583339329-2779392248-2779392248
Content-Length: 334

v=0
o=PortaSIP 2102157303074211757 1 IN IP4 [SipProviderIn]
s=VoipSIP
t=0 0
m=audio 61356 RTP/AVP 18 4 8 0 100
c=IN IP4 [SipProviderIn]
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv

[Oct 10 11:41:19] VERBOSE[24850] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘[ServerIP]’
[Oct 10 11:41:19] VERBOSE[24850] res_pjsip_logger.c: <— Transmitting SIP response (493 bytes) to UDP:[InternalRouterIP]:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [SipProviderOut]:5060;rport=5060;received=[InternalRouterIP];branch=z9hG4bK-524287-1—18dc27db2247d16c477e43d41006afbe
Via: SIP/2.0/UDP [SipProviderIn]:5070;rport=5070;branch=z9hG4bK-5hr4zfmtdhf3reig
Record-Route: sip:SipProviderOut;lr;ep
Call-ID: 0a57cd0f16@41.87.218.2~1o~3o
From: sip:[CallingNumber]@SipProvider;tag=R2b8Jl6YKAndTzPk.o
To: sip:[SipUser]@SipProvider
CSeq: 914 INVITE
Server: Asterisk PBX 16.5.1
Content-Length: 0

[Oct 10 11:41:19] VERBOSE[24855][C-00000006] pbx.c: Executing [[SipUser]@IncomingContext:1] NoOp(“PJSIP/siptrunk-00000006”, "IncomingContext > [SipUser] ") in new stack
[Oct 10 11:41:19] VERBOSE[24855][C-00000006] pbx.c: Executing [[SipUser]@IncomingContext:2] NoOp(“PJSIP/siptrunk-00000006”, "Caller ID: ") in new stack
[Oct 10 11:41:19] VERBOSE[24855][C-00000006] pbx.c: Executing [[SipUser]@IncomingContext:3] Answer(“PJSIP/siptrunk-00000006”, “”) in new stack
[Oct 10 11:41:19] VERBOSE[24850] res_rtp_asterisk.c: 0x55dfc257c320 – Strict RTP learning after remote address set to: [SipProviderIn]:61356
[Oct 10 11:41:19] VERBOSE[24850] res_pjsip_logger.c: <— Transmitting SIP response (988 bytes) to UDP:[InternalRouterIP]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [SipProviderOut]:5060;rport=5060;received=[InternalRouterIP];branch=z9hG4bK-524287-1—18dc27db2247d16c477e43d41006afbe
Via: SIP/2.0/UDP [SipProviderIn]:5070;rport=5070;branch=z9hG4bK-5hr4zfmtdhf3reig
Record-Route: sip:SipProviderOut;lr;ep
Call-ID: 0a57cd0f16@41.87.218.2~1o~3o
From: sip:[CallingNumber]@SipProvider;tag=R2b8Jl6YKAndTzPk.o
To: sip:[SipUser]@SipProvider;tag=9d8d2b4d-9a30-40fa-8b98-9d07a098db92
CSeq: 914 INVITE
Server: Asterisk PBX 16.5.1
Contact: sip:ServerIP:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 230