Hey everyone,
I currently have FreePBX 15 / Asterisk 17 setup and mostly working. Outgoing calls is properly functioning. Calls between extensions also works perfectly. The issue is that Incoming Calls/Origination don’t work. All the communication seems fine, but the FreePBX server is responding to the Twilio server in this case with a “488 Not Acceptable Here”. I’ve checked that I have codecs in common, and that encryption settings are correct. Outgoing calls have no issues.
The only real hint seems to be this error in the Asterisk log:
[2021-04-28 07:16:50] ERROR[1314]: res_pjsip_session.c:934 handle_incoming_sdp: anonymous: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
I’ve left the above error in the request log below so the order of events are preserved.
Here is the pjsip history print out for a summary:
No. Timestamp (Dir) Address SIP Message
===== ========== ============================== ===================================
00000 1619594210 * <== 192.168.201.20:26125 INVITE sip:[email protected];transport=tls;region=us1 SIP/2.0
00001 1619594210 * ==> 192.168.201.20:26125 SIP/2.0 100 Trying
00002 1619594210 * ==> 192.168.201.20:26125 SIP/2.0 488 Not Acceptable Here
00003 1619594210 * <== 192.168.201.20:26125 ACK sip:[email protected];transport=tls;region=us1 SIP/2.0
Here is the logs (I’ve obfuscated some information):
<--- Received SIP request (1596 bytes) from TLS:192.168.201.20:26125 --->
INVITE sip:[email protected];transport=tls;region=us1 SIP/2.0
Record-Route: <sip:54.172.60.0:5061;transport=tls;r2=on;lr>
Record-Route: <sip:54.172.60.0;r2=on;lr>
CSeq: 1 INVITE
From: <sip:[email protected]>;tag=57848935_6772d868_8fcd078b-2979-4af5-af87-4tert535yr3yre
To: <sip:[email protected];transport=tls;region=us1>
Max-Forwards: 65
X-OhSip-Sas-Id: ce52038e-106d-490b-a36d-4g4ff345t
X-OhSIP-Servlet: SipCallOut
X-OhSIP-Remote-Test-Id: sip-call-out_3364
X-OhSIP-Test-Params: {"request_uri":"sip:[email protected]"}
Diversion: <sip:[email protected]>;reason=unconditional
Call-ID: [email protected]
Via: SIP/2.0/TLS 54.172.60.0:5061;branch=z9hG4bK04b6.a4deb99b323fdsdf4g522d3a31e98635.0
Via: SIP/2.0/UDP 172.25.89.198:5060;rport=5060;branch=z9hG4bK8fcd078b-2979-4af5-af87-a64743721673_6772d868_431-10982788592345641489
Contact: <sip:[email protected]:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY
User-Agent: Twilio Gateway
X-Twilio-AccountSid: ACe03671ace6f4f849dfe5c363c6246545
Content-Type: application/sdp
X-Twilio-CallSid: CAe79a965bfbe6bd4ab67de524719cd8765
Content-Length: 361
v=0
o=root 309348332 309348332 IN IP4 172.18.146.83
s=Twilio Media Gateway
c=IN IP4 34.203.251.159
t=0 0
m=audio 10244 RTP/SAVP 0 8 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HboccKn2v8m3QR6RaNAoZ2LtMFslAIk0DFVC342
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
<--- Transmitting SIP response (657 bytes) to TLS:192.168.201.20:26125 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 54.172.60.0:5061;rport=26125;received=192.168.201.20;branch=z9hG4bK04b6.a4deb99b323fdsdf4g522d3a31e98635.0
Via: SIP/2.0/UDP 172.25.89.198:5060;rport=5060;branch=z9hG4bK8fcd078b-2979-4af5-af87-a64743721673_6772d868_431-10982788592345641489
Record-Route: <sip:54.172.60.0:5061;transport=tls;lr;r2=on>
Record-Route: <sip:54.172.60.0;lr;r2=on>
Call-ID: [email protected]
From: <sip:[email protected]>;tag=57848935_6772d868_8fcd078b-2979-4af5-af87-a64743721673
To: <sip:[email protected];region=us1>
CSeq: 1 INVITE
Server: FPBX-15.0.17.32(17.9.3)
Content-Length: 0
[2021-04-28 07:16:50] ERROR[1314]: res_pjsip_session.c:934 handle_incoming_sdp: anonymous: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
<--- Transmitting SIP response (711 bytes) to TLS:192.168.201.20:26125 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 54.172.60.0:5061;rport=26125;received=192.168.201.20;branch=z9hG4bK04b6.a4deb99b323fdsdf4g522d3a31e98635.0
Via: SIP/2.0/UDP 172.25.89.198:5060;rport=5060;branch=z9hG4bK8fcd078b-2979-4af5-af87-a64743721673_6772d868_431-10982788592345641489
Record-Route: <sip:54.172.60.0:5061;transport=tls;lr;r2=on>
Record-Route: <sip:54.172.60.0;lr;r2=on>
Call-ID: [email protected]
From: <sip:[email protected]>;tag=57848935_6772d868_8fcd078b-2979-4af5-af87-a64743721673
To: <sip:[email protected];region=us1>;tag=c84eb84f-9160-4989-b101-81fbc1a4c695
CSeq: 1 INVITE
Server: FPBX-15.0.17.32(17.9.3)
Content-Length: 0
<--- Received SIP request (457 bytes) from TLS:192.168.201.20:26125 --->
ACK sip:[email protected];transport=tls;region=us1 SIP/2.0
CSeq: 1 ACK
From: <sip:[email protected]>;tag=57848935_6772d868_8fcd078b-2979-4af5-af87-a64743721673
To: <sip:[email protected];region=us1>;tag=c84eb84f-9160-4989-b101-81fbc1a4c695
Max-Forwards: 65
Call-ID: [email protected]
Via: SIP/2.0/TLS 54.172.60.0:5061;branch=z9hG4bK04b6.a4deb99b323fdsdf4g522d3a31e98635.0
Content-Length: 0
Any help would be greatly appreciated!