I’m not sure how I failed to paste the full log the first time, here is a second attempt.
<--- Received SIP request (970 bytes) from UDP:172.19.12.116:5060 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 172.19.12.116;branch=z9hG4bKce84d1003E77B5C7
From: "3065" <sip:[email protected]>;tag=328613B6-709D3375
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
Call-ID: [email protected]
Contact: <sip:[email protected]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_311-UA/5.4.5.8201
Accept-Language: en
Supported: replaces,100rel
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 352
v=0
o=- 1539358599 1539358599 IN IP4 172.19.12.116
s=Polycom IP Phone
c=IN IP4 172.19.12.116
t=0 0
a=sendrecv
m=audio 2248 RTP/AVP 9 102 0 8 18 127
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
<--- Transmitting SIP response (521 bytes) to UDP:172.19.12.116:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.19.12.116;rport=5060;received=172.19.12.116;branch=z9hG4bKce84d1003E77B5C7
Call-ID: [email protected]
From: "3065" <sip:[email protected]>;tag=328613B6-709D3375
To: <sip:[email protected];user=phone>;tag=z9hG4bKce84d1003E77B5C7
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1539358600/e5e82cdb410b8e97d08580dc176610d8",opaque="4858559f62da2599",algorithm=md5,qop="auth"
Server: FPBX-14.0.3.19(15.5.0)
Content-Length: 0
<--- Received SIP request (543 bytes) from UDP:172.19.12.116:5060 --->
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 172.19.12.116;branch=z9hG4bKce84d1003E77B5C7
From: "3065" <sip:[email protected]>;tag=328613B6-709D3375
To: <sip:[email protected];user=phone>;tag=z9hG4bKce84d1003E77B5C7
CSeq: 1 ACK
Call-ID: [email protected]
Contact: <sip:[email protected]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_311-UA/5.4.5.8201
Accept-Language: en
Max-Forwards: 70
Content-Length: 0
<--- Received SIP request (1257 bytes) from UDP:172.19.12.116:5060 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 172.19.12.116;branch=z9hG4bK7715953b1D8A0CFE
From: "3065" <sip:[email protected]>;tag=328613B6-709D3375
To: <sip:[email protected];user=phone>
CSeq: 2 INVITE
Call-ID: [email protected]
Contact: <sip:[email protected]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_311-UA/5.4.5.8201
Accept-Language: en
Supported: replaces,100rel
Allow-Events: conference,talk,hold
Authorization: Digest username="3065", realm="asterisk", nonce="1539358600/e5e82cdb410b8e97d08580dc176610d8", qop=auth, cnonce="5ytzULxhgvpKs4p", nc=00000001, opaque="4858559f62da2599", uri="sip:[email protected];user=phone", response="a38ff2ecf25ca58f941b179ec36a9935", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 352
v=0
o=- 1539358599 1539358599 IN IP4 172.19.12.116
s=Polycom IP Phone
c=IN IP4 172.19.12.116
t=0 0
a=sendrecv
m=audio 2248 RTP/AVP 9 102 0 8 18 127
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
<--- Transmitting SIP response (340 bytes) to UDP:172.19.12.116:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.19.12.116;rport=5060;received=172.19.12.116;branch=z9hG4bK7715953b1D8A0CFE
Call-ID: [email protected]
From: "3065" <sip:[email protected]>;tag=328613B6-709D3375
To: <sip:[email protected];user=phone>
CSeq: 2 INVITE
Server: FPBX-14.0.3.19(15.5.0)
Content-Length: 0
[2018-10-12 10:36:40] ERROR[18744]: res_pjsip.c:3295 ast_sip_create_dialog_uac: Endpoint 'CIC': Could not create dialog to invalid URI 'CIC'. Is endpoint registered and reachable?
[2018-10-12 10:36:40] ERROR[18744]: chan_pjsip.c:2497 request: Failed to create outgoing session to endpoint 'CIC'
[2018-10-12 10:36:40] WARNING[22252][C-0000000e]: app_dial.c:2512 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
<--- Transmitting SIP response (927 bytes) to UDP:172.19.12.116:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.19.12.116;rport=5060;received=172.19.12.116;branch=z9hG4bK7715953b1D8A0CFE
Call-ID: [email protected]
From: "3065" <sip:[email protected]>;tag=328613B6-709D3375
To: <sip:[email protected];user=phone>;tag=4fcac243-6dae-499a-b762-47189915b4c0
CSeq: 2 INVITE
Server: FPBX-14.0.3.19(15.5.0)
Contact: <sip:172.19.11.206:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: "CID:3065" <sip:[email protected];user=phone>
Content-Type: application/sdp
Content-Length: 287
v=0
o=- 1539358599 1539358601 IN IP4 172.19.11.206
s=Asterisk
c=IN IP4 172.19.11.206
t=0 0
m=audio 18424 RTP/AVP 0 8 9 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=ptime:20
a=maxptime:150
a=sendrecv