Here’s the output:
<— SIP read from TCP:S4B IP:62176 —>
INVITE sip:4000@freepbx.domain;user=phone SIP/2.0
FROM: “testuser”<sip:0456@ domain;user=phone>;epid=1A2512F9FC;tag=18a78fc448
TO: <sip:4000@ freepbx.domain;user=phone>
CSEQ: 4523 INVITE
CALL-ID: d627a246-603e-4973-8bb3-2f18db70bf60
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP S4B IP:62176;branch=z9hG4bKd315c5f
CONTACT: <sip:S4B IP.domain:5060;transport=Tcp;maddr=S4B IP;ms-opaque=2bd72463f49d1de5>
CONTENT-LENGTH: 336
SUPPORTED: 100rel
USER-AGENT: RTCC/7.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
v=0
o=- 74 1 IN IP4 S4B IP
s=session
c=IN IP4 S4B IP
b=CT:1000
t=0 0
m=audio 56668 RTP/AVP 97 101 13 0 8
c=IN IP4 S4B IP
a=rtcp:56669
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
— (14 headers 18 lines) —
Sending to S4B IP:62176 (no NAT)
Sending to S4B IP:62176 (no NAT)
Using INVITE request as basis request - d627a246-603e-4973-8bb3-2f18db70bf60
No matching peer for ‘0456’ from ‘S4B IP:62176’
Found RTP audio format 97
Found RTP audio format 101
Found RTP audio format 13
Found RTP audio format 0
Found RTP audio format 8
Found unknown media description format RED for ID 97
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 13
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port S4B IP:56668
Looking for 4000 in from-sip-external (domain freepbx.domain)
sip_route_dump: route/path hop: <sip:S4B IP.domain:5060;transport=Tcp;maddr=S4B IP;ms-opaque=2bd72463f49d1de5>
<— Transmitting (no NAT) to S4B IP:62176 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP S4B IP:62176;branch=z9hG4bKd315c5f;received=S4B IP
From: “testuser”<sip:0456@ domain;user=phone>;epid=1A2512F9FC;tag=18a78fc448
To: <sip:4000@ freepbx.domain;user=phone>
Call-ID: d627a246-603e-4973-8bb3-2f18db70bf60
CSeq: 4523 INVITE
Server: FPBX-14.0.13.4(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4000@SIP Phone IP:5060;transport=tcp>
Content-Length: 0
<------------>
[2019-07-29 12:10:17] WARNING[11180][C-00000002]: Ext. s:6 @ from-sip-external: “Rejecting unknown SIP connection from S4B IP”
Audio is at 10008
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (no NAT) to S4B IP:62176 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP S4B client IP:62176;branch=z9hG4bKd315c5f;received=S4B client IP
From: “testuser”<sip:0456@ domain;user=phone>;epid=1A2512F9FC;tag=18a78fc448
To: <sip:4000@ freepbx.domain;user=phone>;tag=as638452aa
Call-ID: d627a246-603e-4973-8bb3-2f18db70bf60
CSeq: 4523 INVITE
Server: FPBX-14.0.13.4(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4000@FREEPBX IP:5060;transport=tcp>
Content-Type: application/sdp
Content-Length: 274
v=0
o=root 1635555308 1635555308 IN IP4 FreePBX IP
s=Asterisk PBX 13.22.0
c=IN IP4 FreePBX IP
t=0 0
m=audio 10008 RTP/AVP 0 8 101
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:150
a=sendrecv
<------------>
<— SIP read from TCP:SIP Phone IP:11844 —>
REGISTER sip:freepbx.domain:5060 SIP/2.0
Via: SIP/2.0/TCP SIP Phone IP:5060;branch=z9hG4bK958848065
From: “3000” <sip:3000@ freepbx.domain:5060>;tag=2404835808
To: “3000” <sip:3000@ freepbx.domain:5060>
Call-ID: 0_2956647971@SIP IP Phone
CSeq: 91 REGISTER
Contact: <sip:3000@SIP Phone IP:5060;transport=TCP>
Authorization: Digest username=“3000”, realm=“asterisk”, nonce=“66419d8a”, uri=“sip:freepbx.domain:5060”, response=“578205542d3652b507594c3f8935a134”, algorithm=MD5
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink W60B 77.83.0.25
Expires: 180
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Sending to SIP Phone IP:5060 (no NAT)
Sending to SIP Phone IP:5060 (no NAT)
<— Transmitting (no NAT) to 10.50.2.109:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP SIP Phone IP:5060;branch=z9hG4bK958848065;received=10.50.2.109
From: “3000” <sip:3000@ freepbx.domain:5060>;tag=2404835808
To: “3000” <sip:3000@ freepbx.domain:5060>;tag=as01cceb98
Call-ID: 0_2956647971@SIP IP Phone
CSeq: 91 REGISTER
Server: FPBX-14.0.13.4(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“66310148”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0_2956647971@SIP Phone IP’ in 32000 ms (Method: REGISTER)
<— SIP read from TCP: SIP Phone IP:11844 —>
REGISTER sip:freepbx.domain:5060 SIP/2.0
Via: SIP/2.0/TCP SIP Phone IP:5060;branch=z9hG4bK1463243007
From: “3000” <sip:3000@ freepbx.domain:5060>;tag=2404835808
To: “3000” <sip:3000@ freepbx.domain:5060>
Call-ID: 0_2956647971@ SIP IP Phone
CSeq: 92 REGISTER
Contact: <sip:3000@SIP Phone IP:5060;transport=TCP>
Authorization: Digest username=“3000”, realm=“asterisk”, nonce=“66310148”, uri=“sip:freepbx.domain:5060”, response=“5a0a8133a1358c642bce97208ae8ba60”, algorithm=MD5
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink W60B 77.83.0.25
Expires: 180
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Sending to SIP Phone IP:5060 (no NAT)
Reliably Transmitting (no NAT) to SIP Phone IP:5060:
OPTIONS sip:3000@SIP Phone IP:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP FreePBX IP:5060;branch=z9hG4bK23ec7a52
Max-Forwards: 70
From: “Unknown” <sip:Unknown@FreePBX IP>;tag=as0c5c1cd1
To: <sip:3000@SIP Phone IP:5060;transport=TCP>
Contact: <sip:Unknown@FreePBX IP:5060;transport=tcp>
Call-ID: 251a842643448a816502125c563e9b9f@FreePBX IP:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.22.0)
Date: Mon, 29 Jul 2019 16:10:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— Transmitting (no NAT) to SIP Phone IP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP SIP Phone IP:5060;branch=z9hG4bK1463243007;received=SIP Phone IP
From: “3000” <sip:3000@ freepbx.domain:5060>;tag=2404835808
To: “3000” <sip:3000@ freepbx.domain:5060>;tag=as01cceb98
Call-ID: 0_2956647971@SIP Phone IP
CSeq: 92 REGISTER
Server: FPBX-14.0.13.4(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 180
Contact: <sip:3000@SIP Phone IP:5060;transport=TCP>;expires=180
Date: Mon, 29 Jul 2019 16:10:17 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0_2956647971@SIP Phone’ in 32000 ms (Method: REGISTER)
<— SIP read from TCP:SIP Phone:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP FreePBX IP:5060;branch=z9hG4bK23ec7a52
From: “Unknown” <sip:Unknown@FreePBX IP>;tag=as0c5c1cd1
To: <sip:3000@SIP phone:5060;transport=TCP>;tag=3094584608
Call-ID: 251a842643448a816502125c563e9b9f@FreePBX IP:5060
CSeq: 102 OPTIONS
User-Agent: Yealink W60B 77.83.0.25
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from TCP: S4B IP:62176 —>
ACK sip:4000@FreePBX IP:5060;transport=tcp SIP/2.0
FROM: <sip:0456@ domain;user=phone>;epid=1A2512F9FC;tag=18a78fc448
TO: <sip:4000@ freepbx.domain;user=phone>;tag=as638452aa
CSEQ: 4523 ACK
CALL-ID: d627a246-603e-4973-8bb3-2f18db70bf60
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP S4B IP:62176;branch=z9hG4bKbbf4eb81
CONTENT-LENGTH: 0
USER-AGENT: RTCC/7.0.0.0 MediationServer
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘251a842643448a816502125c563e9b9f@FreePBX IP:5060’ Method: OPTIONS
<— SIP read from TCP:SIP phone:11844 —>
<------------->
<— SIP read from TCP: S4B IP:62176 —>
BYE sip:4000@ S4B IP:5060;transport=tcp SIP/2.0
FROM: <sip:0456@ domain;user=phone>;epid=1A2512F9FC;tag=18a78fc448
TO: <sip:4000@ freepbxdomain;user=phone>;tag=as638452aa
CSEQ: 4524 BYE
CALL-ID: d627a246-603e-4973-8bb3-2f18db70bf60
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP S4B IP:62176;branch=z9hG4bKdb834c9
CONTACT: <sip: S4B IP.domain:5060;transport=Tcp;maddr= S4B IP>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/7.0.0.0 MediationServer
<------------->
— (10 headers 0 lines) —
Sending to S4B IP:62176 (no NAT)
Scheduling destruction of SIP dialog ‘d627a246-603e-4973-8bb3-2f18db70bf60’ in 32000 ms (Method: BYE)
<— Transmitting (no NAT) to S4B IP:62176 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP S4B IP:62176;branch=z9hG4bKdb834c9;received=S4B IP
From: <sip:0456@ domain;user=phone>;epid=1A2512F9FC;tag=18a78fc448
To: <sip:4000@ freepbx.domain;user=phone>;tag=as638452aa
Call-ID: d627a246-603e-4973-8bb3-2f18db70bf60
CSeq: 4524 BYE
Server: FPBX-14.0.13.4(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog ‘e0373861b0cf4add8e26c383bfa295b4’ Method: OPTIONS