Oh right sorry here is sngrep with timings
timer1 = 500
timerb = 32000
second leg is ~150ms away
<------------->
[Apr 22 09:58:00] VERBOSE[19959] chan_sip.c: --- (13 headers 0 lines) ---
[Apr 22 09:58:00] VERBOSE[19959] chan_sip.c: Really destroying SIP dialog '22b21cf5136d19f84ee04ce4001e0cf5@10.102.0.143:5060' Method: OPTIONS
[2022-04-22 09:58:04.719] Asterisk 18.11.1 built by root @ asterisk-ubuntu-new on a x86_64 running Linux on 2022-04-06 06:50:03 UTC
[Apr 22 09:58:04] VERBOSE[10053] loader.c: Reloading module 'logger' (Logger)
[2022-04-22 09:58:04.719] VERBOSE[10053] logger.c: Asterisk Queue Logger restarted
[2022-04-22 09:58:05.314] VERBOSE[10053] asterisk.c: Remote UNIX connection disconnected
[2022-04-22 09:58:18.666] VERBOSE[19959][C-00000017] netsock2.c: Using SIP RTP CoS mark 5
[2022-04-22 09:58:18.667] VERBOSE[19959][C-00000017] res_rtp_asterisk.c: 0x7f84f00651a0 -- Strict RTP learning after remote address set to: 10.100.255.98:4006
[2022-04-22 09:58:18.667] NOTICE[19959] chan_sip.c: Still have a QUALIFY dialog active, deleting
[2022-04-22 09:58:18.668] VERBOSE[10057][C-00000017] pbx.c: Executing [123@access-level3:1] Dial("SIP/200700-0000001e", "SIP/helpdesk-ubuntu/422125") in new stack
[2022-04-22 09:58:18.669] VERBOSE[10057][C-00000017] netsock2.c: Using SIP RTP CoS mark 5
[2022-04-22 09:58:18.670] VERBOSE[10057][C-00000017] chan_sip.c: Audio is at 18228
[2022-04-22 09:58:18.670] VERBOSE[10057][C-00000017] chan_sip.c: Adding codec alaw to SDP
[2022-04-22 09:58:18.670] VERBOSE[10057][C-00000017] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2022-04-22 09:58:18.671] VERBOSE[10057][C-00000017] chan_sip.c: Reliably Transmitting (no NAT) to 10.102.96.95:5060:
INVITE sip:422125@10.102.96.95:5060 SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK0ec77795
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>
Contact: <sip:200700@10.102.0.143:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 102 INVITE
User-Agent: Test PABX AU
Date: Thu, 21 Apr 2022 23:58:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 381049237 381049237 IN IP4 10.102.0.143
s=Test PABX AU
c=IN IP4 10.100.255.98
t=0 0
m=audio 4006 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
[2022-04-22 09:58:18.673] VERBOSE[10057][C-00000017] app_dial.c: Called SIP/helpdesk-ubuntu/422125
[2022-04-22 09:58:18.822] VERBOSE[19959] chan_sip.c: Retransmitting #1 (no NAT) to 10.102.96.95:5060:
INVITE sip:422125@10.102.96.95:5060 SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK0ec77795
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>
Contact: <sip:200700@10.102.0.143:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 102 INVITE
User-Agent: Test PABX AU
Date: Thu, 21 Apr 2022 23:58:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 381049237 381049237 IN IP4 10.102.0.143
s=Test PABX AU
c=IN IP4 10.100.255.98
t=0 0
m=audio 4006 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
[2022-04-22 09:58:18.825] VERBOSE[19959] chan_sip.c:
<--- SIP read from UDP:10.102.96.95:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK0ec77795
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 102 INVITE
User-Agent: FreeSWITCH-Test
Content-Length: 0
<------------->
[2022-04-22 09:58:18.826] VERBOSE[19959] chan_sip.c: --- (8 headers 0 lines) ---
[2022-04-22 09:58:18.828] VERBOSE[19959] chan_sip.c:
<--- SIP read from UDP:10.102.96.95:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK0ec77795
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 102 INVITE
Contact: <sip:422125@10.102.96.95:5060;transport=udp>
User-Agent: FreeSWITCH-Test
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
Remote-Party-ID: "422125" <sip:422125@10.102.96.95>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1650572442 1650572443 IN IP4 10.102.96.95
s=FreeSWITCH
c=IN IP4 10.102.96.95
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
<------------->
[2022-04-22 09:58:18.829] VERBOSE[19959] chan_sip.c: --- (16 headers 11 lines) ---
[2022-04-22 09:58:18.829] VERBOSE[19959][C-00000017] chan_sip.c: Got SDP version 1650572443 and unique parts [FreeSWITCH 1650572442 IN IP4 10.102.96.95]
[2022-04-22 09:58:18.830] VERBOSE[19959][C-00000017] chan_sip.c: Found RTP audio format 8
[2022-04-22 09:58:18.830] VERBOSE[19959][C-00000017] chan_sip.c: Found RTP audio format 101
[2022-04-22 09:58:18.830] VERBOSE[19959][C-00000017] chan_sip.c: Found audio description format PCMA for ID 8
[2022-04-22 09:58:18.830] VERBOSE[19959][C-00000017] chan_sip.c: Found audio description format telephone-event for ID 101
[2022-04-22 09:58:18.830] VERBOSE[19959][C-00000017] chan_sip.c: Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[2022-04-22 09:58:18.830] VERBOSE[19959][C-00000017] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2022-04-22 09:58:18.830] VERBOSE[19959][C-00000017] res_rtp_asterisk.c: 0x7f8584015be0 -- Strict RTP learning after remote address set to: 10.102.96.95:13056
[2022-04-22 09:58:18.831] VERBOSE[19959][C-00000017] chan_sip.c: Peer audio RTP is at port 10.102.96.95:13056
[2022-04-22 09:58:18.831] VERBOSE[19959][C-00000017] sip/route.c: sip_route_dump: route/path hop: <sip:422125@10.102.96.95:5060;transport=udp>
[2022-04-22 09:58:18.831] VERBOSE[19959][C-00000017] chan_sip.c: set_destination: Parsing <sip:422125@10.102.96.95:5060;transport=udp> for address/port to send to
[2022-04-22 09:58:18.831] VERBOSE[19959][C-00000017] chan_sip.c: set_destination: set destination to 10.102.96.95:5060
[2022-04-22 09:58:18.831] VERBOSE[19959][C-00000017] chan_sip.c: Transmitting (no NAT) to 10.102.96.95:5060:
ACK sip:422125@10.102.96.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK2ebd81eb
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Contact: <sip:200700@10.102.0.143:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 102 ACK
User-Agent: Test PABX AU
Content-Length: 0
---
[2022-04-22 09:58:18.832] VERBOSE[10057][C-00000017] app_dial.c: SIP/helpdesk-ubuntu-0000001f answered SIP/200700-0000001e
[2022-04-22 09:58:18.833] VERBOSE[10058][C-00000017] bridge_channel.c: Channel SIP/helpdesk-ubuntu-0000001f joined 'simple_bridge' basic-bridge <fe189437-1765-4eb0-bd56-4d8f2ad72e90>
[2022-04-22 09:58:18.833] VERBOSE[10057][C-00000017] bridge_channel.c: Channel SIP/200700-0000001e joined 'simple_bridge' basic-bridge <fe189437-1765-4eb0-bd56-4d8f2ad72e90>
[2022-04-22 09:58:18.834] VERBOSE[10057][C-00000017] bridge.c: Bridge fe189437-1765-4eb0-bd56-4d8f2ad72e90: switching from simple_bridge technology to native_rtp
[2022-04-22 09:58:18.834] VERBOSE[10057][C-00000017] bridge_native_rtp.c: Remotely bridged 'SIP/200700-0000001e' and 'SIP/helpdesk-ubuntu-0000001f' - media will flow directly between them
[2022-04-22 09:58:18.972] VERBOSE[19959] chan_sip.c:
<--- SIP read from UDP:10.102.96.95:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK0ec77795
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 102 INVITE
Contact: <sip:422125@10.102.96.95:5060;transport=udp>
User-Agent: FreeSWITCH-Test
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
Remote-Party-ID: "422125" <sip:422125@10.102.96.95>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1650572442 1650572443 IN IP4 10.102.96.95
s=FreeSWITCH
c=IN IP4 10.102.96.95
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
<------------->
[2022-04-22 09:58:18.973] VERBOSE[19959] chan_sip.c: --- (16 headers 11 lines) ---
[2022-04-22 09:58:18.973] VERBOSE[19959][C-00000017] chan_sip.c: set_destination: Parsing <sip:422125@10.102.96.95:5060;transport=udp> for address/port to send to
[2022-04-22 09:58:18.973] VERBOSE[19959][C-00000017] chan_sip.c: set_destination: set destination to 10.102.96.95:5060
[2022-04-22 09:58:18.973] VERBOSE[19959][C-00000017] chan_sip.c: Transmitting (no NAT) to 10.102.96.95:5060:
ACK sip:422125@10.102.96.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK26485ab4
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Contact: <sip:200700@10.102.0.143:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 102 ACK
User-Agent: Test PABX AU
Content-Length: 0
---
[2022-04-22 09:58:21.609] VERBOSE[10057][C-00000017] bridge_channel.c: Channel SIP/200700-0000001e left 'native_rtp' basic-bridge <fe189437-1765-4eb0-bd56-4d8f2ad72e90>
[2022-04-22 09:58:21.609] VERBOSE[10057][C-00000017] chan_sip.c: set_destination: Parsing <sip:422125@10.102.96.95:5060;transport=udp> for address/port to send to
[2022-04-22 09:58:21.609] VERBOSE[10057][C-00000017] chan_sip.c: set_destination: set destination to 10.102.96.95:5060
[2022-04-22 09:58:21.609] VERBOSE[10057][C-00000017] chan_sip.c: Audio is at 18228
[2022-04-22 09:58:21.609] VERBOSE[10057][C-00000017] chan_sip.c: Adding codec alaw to SDP
[2022-04-22 09:58:21.609] VERBOSE[10057][C-00000017] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2022-04-22 09:58:21.609] VERBOSE[10057][C-00000017] chan_sip.c: Reliably Transmitting (no NAT) to 10.102.96.95:5060:
INVITE sip:422125@10.102.96.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK29b83c62
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Contact: <sip:200700@10.102.0.143:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 103 INVITE
User-Agent: Test PABX AU
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 381049237 381049238 IN IP4 10.102.0.143
s=Test PABX AU
c=IN IP4 10.102.0.143
t=0 0
m=audio 18228 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
[2022-04-22 09:58:21.609] VERBOSE[10058][C-00000017] bridge_channel.c: Channel SIP/helpdesk-ubuntu-0000001f left 'native_rtp' basic-bridge <fe189437-1765-4eb0-bd56-4d8f2ad72e90>
[2022-04-22 09:58:21.610] VERBOSE[10057][C-00000017] pbx.c: Spawn extension (access-level3, 123, 1) exited non-zero on 'SIP/200700-0000001e'
[2022-04-22 09:58:21.610] VERBOSE[10058][C-00000017] chan_sip.c: Scheduling destruction of SIP dialog '3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060' in 9600 ms (Method: INVITE)
[2022-04-22 09:58:21.759] VERBOSE[19959] chan_sip.c: Retransmitting #1 (no NAT) to 10.102.96.95:5060:
INVITE sip:422125@10.102.96.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK29b83c62
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Contact: <sip:200700@10.102.0.143:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 103 INVITE
User-Agent: Test PABX AU
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 381049237 381049238 IN IP4 10.102.0.143
s=Test PABX AU
c=IN IP4 10.102.0.143
t=0 0
m=audio 18228 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
[2022-04-22 09:58:21.776] VERBOSE[19959] chan_sip.c:
<--- SIP read from UDP:10.102.96.95:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK29b83c62
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 103 INVITE
User-Agent: FreeSWITCH-Test
Content-Length: 0
<------------->
[2022-04-22 09:58:21.777] VERBOSE[19959] chan_sip.c: --- (8 headers 0 lines) ---
[2022-04-22 09:58:21.778] VERBOSE[19959] chan_sip.c:
<--- SIP read from UDP:10.102.96.95:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK29b83c62
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 103 INVITE
Contact: <sip:422125@10.102.96.95:5060;transport=udp>
User-Agent: FreeSWITCH-Test
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
v=0
o=FreeSWITCH 1650572442 1650572443 IN IP4 10.102.96.95
s=FreeSWITCH
c=IN IP4 10.102.96.95
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
<------------->
[2022-04-22 09:58:21.780] VERBOSE[19959] chan_sip.c: --- (14 headers 11 lines) ---
[2022-04-22 09:58:21.780] VERBOSE[19959][C-00000017] chan_sip.c: Comparing SDP version 1650572443 -> 1650572443 and unique parts [FreeSWITCH 1650572442 IN IP4 10.102.96.95] -> [FreeSWITCH 1650572442 IN IP4 10.102.96.95]
[2022-04-22 09:58:21.780] VERBOSE[19959][C-00000017] chan_sip.c: set_destination: Parsing <sip:422125@10.102.96.95:5060;transport=udp> for address/port to send to
[2022-04-22 09:58:21.780] VERBOSE[19959][C-00000017] chan_sip.c: set_destination: set destination to 10.102.96.95:5060
[2022-04-22 09:58:21.780] VERBOSE[19959][C-00000017] chan_sip.c: Transmitting (no NAT) to 10.102.96.95:5060:
ACK sip:422125@10.102.96.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK3558d59c
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Contact: <sip:200700@10.102.0.143:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 103 ACK
User-Agent: Test PABX AU
Content-Length: 0
---
[2022-04-22 09:58:21.781] VERBOSE[19959] chan_sip.c: set_destination: Parsing <sip:422125@10.102.96.95:5060;transport=udp> for address/port to send to
[2022-04-22 09:58:21.781] VERBOSE[19959] chan_sip.c: set_destination: set destination to 10.102.96.95:5060
[2022-04-22 09:58:21.781] VERBOSE[19959] chan_sip.c: Reliably Transmitting (no NAT) to 10.102.96.95:5060:
BYE sip:422125@10.102.96.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK47cdf1ab
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 104 BYE
User-Agent: Test PABX AU
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[2022-04-22 09:58:21.781] VERBOSE[19959] chan_sip.c: Scheduling destruction of SIP dialog '3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060' in 9600 ms (Method: INVITE)
[2022-04-22 09:58:21.909] VERBOSE[19959] chan_sip.c:
<--- SIP read from UDP:10.102.96.95:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK29b83c62
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 103 INVITE
Contact: <sip:422125@10.102.96.95:5060;transport=udp>
User-Agent: FreeSWITCH-Test
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
v=0
o=FreeSWITCH 1650572442 1650572443 IN IP4 10.102.96.95
s=FreeSWITCH
c=IN IP4 10.102.96.95
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
<------------->
[2022-04-22 09:58:21.909] VERBOSE[19959] chan_sip.c: --- (14 headers 11 lines) ---
[2022-04-22 09:58:21.909] VERBOSE[19959][C-00000017] chan_sip.c: set_destination: Parsing <sip:422125@10.102.96.95:5060;transport=udp> for address/port to send to
[2022-04-22 09:58:21.909] VERBOSE[19959][C-00000017] chan_sip.c: set_destination: set destination to 10.102.96.95:5060
[2022-04-22 09:58:21.909] VERBOSE[19959][C-00000017] chan_sip.c: Transmitting (no NAT) to 10.102.96.95:5060:
ACK sip:422125@10.102.96.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK23ea07be
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Contact: <sip:200700@10.102.0.143:5060>
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 103 ACK
User-Agent: Test PABX AU
Content-Length: 0
---
[2022-04-22 09:58:21.930] VERBOSE[19959] chan_sip.c: Retransmitting #1 (no NAT) to 10.102.96.95:5060:
BYE sip:422125@10.102.96.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK47cdf1ab
Max-Forwards: 70
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 104 BYE
User-Agent: Test PABX AU
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[2022-04-22 09:58:21.937] VERBOSE[19959] chan_sip.c:
<--- SIP read from UDP:10.102.96.95:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK47cdf1ab
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 104 BYE
User-Agent: FreeSWITCH-Test
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0
<------------->
[2022-04-22 09:58:21.938] VERBOSE[19959] chan_sip.c: --- (10 headers 0 lines) ---
[2022-04-22 09:58:21.938] VERBOSE[19959] chan_sip.c: Really destroying SIP dialog '3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060' Method: INVITE
[2022-04-22 09:58:22.080] VERBOSE[19959] chan_sip.c:
<--- SIP read from UDP:10.102.96.95:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.102.0.143:5060;branch=z9hG4bK47cdf1ab
From: "Daniel Smith Test" <sip:200700@10.102.0.143>;tag=as215a4017
To: <sip:422125@10.102.96.95:5060>;tag=714v0Nvyt00Sa
Call-ID: 3e3bcecf080f86ba35d659043b9ddeec@10.102.0.143:5060
CSeq: 104 BYE
User-Agent: FreeSWITCH-Test
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0
<------------->