So here is our situation. We are using SIPp to test some call flows. The call flows involve screening and leverage an auto-attendant that is leveraging ari4Java. When we have screened the callee (transfer target) we do a refer with the ARI. We are able to get all the steps outlined in the RFC to work. However, after the last BYE/200 from transferor to transferee about 34 seconds elapse and then Asterisk (transferor) issues a SUBSCRIBE to the transferee. This is not illustrated in the RFC and we believe they should not be happening. The SUBSCRIBEs seem to be related directly to the refer because the Event header says “refer”. I am assuming there is some config I am missing or something simple and hoping you can help. Here is the entire log for the call flow from asterisk. I left the sip messages in-tact but deleted some logs before the refer and some duplicate SUBSCRIBEs at the end so I could get below 32k characters, otherwise the websit blocks me.
<— Received SIP request (570 bytes) from UDP:192.22.33.42:5060 —>
INVITE sip:19995551212@192.22.33.188:5060 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.42:5060;branch=z9hG4bK-650-1-0
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188
Call-ID: 1-650@192.17.0.2
CSeq: 1 INVITE
Contact: “SIPpTest” sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 123
v=0
o=19995551313 0 0 IN IP4 192.22.33.42
s=-
c=IN IP4 192.17.0.2
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<— Transmitting SIP response (471 bytes) to UDP:192.22.33.42:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.22.33.42:5060;rport=5060;received=192.22.33.42;branch=z9hG4bK-650-1-0
Call-ID: 1-650@192.17.0.2
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188;tag=z9hG4bK-650-1-0
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1523032815/fe49c999a305e631534ab1dbbb95f8bd”,opaque=“164da8996bc530ae”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 15.1.2
Content-Length: 0
<— Received SIP request (307 bytes) from UDP:192.22.33.42:5060 —>
ACK sip:19995551212@192.22.33.188:5060 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.42:5060
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188:5060;tag=z9hG4bK-650-1-0
Call-ID: 1-650@192.17.0.2
CSeq: 1 ACK
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
<— Received SIP request (838 bytes) from UDP:192.22.33.42:5060 —>
INVITE sip:19995551212@192.22.33.188:5060 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.42:5060;branch=z9hG4bK-650-1-3
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188
Call-ID: 1-650@192.17.0.2
CSeq: 2 INVITE
Contact: “SIPpTest” sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188
Authorization: Digest username=“19995551313”,realm=“asterisk”,cnonce=“6b8b4567”,nc=00000001,qop=auth,uri=“sip:192.22.33.188:5060”,nonce=“1523032815/fe49c999a305e631534ab1dbbb95f8bd”,response=“8fdf728d018307c9f8ec312b039f7a94”,algorithm=md5,opaque=“164da8996bc530ae”
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 123
v=0
o=19995551313 0 0 IN IP4 192.22.33.42
s=-
c=IN IP4 192.17.0.2
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
== Setting global variable ‘SIPDOMAIN’ to ‘192.22.33.188’
<— Transmitting SIP response (298 bytes) to UDP:192.22.33.42:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.22.33.42:5060;rport=5060;received=192.22.33.42;branch=z9hG4bK-650-1-3
Call-ID: 1-650@192.17.0.2
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188
CSeq: 2 INVITE
Server: Asterisk PBX 15.1.2
Content-Length: 0
<— Transmitting SIP response (728 bytes) to UDP:192.22.33.42:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.22.33.42:5060;rport=5060;received=192.22.33.42;branch=z9hG4bK-650-1-3
Call-ID: 1-650@192.17.0.2
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
CSeq: 2 INVITE
Server: Asterisk PBX 15.1.2
Contact: sip:192.22.33.188:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 165
v=0
o=- 0 2 IN IP4 192.22.33.188
s=Asterisk
c=IN IP4 192.22.33.188
t=0 0
m=audio 20512 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv
<— Received SIP request (702 bytes) from UDP:192.22.33.42:5060 —>
ACK sip:192.22.33.188:5060 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.42:5060;branch=z9hG4bK-650-1-7
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.214;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
Call-ID: 1-650@192.17.0.2
CSeq: 2 ACK
Contact: “SIPpTest” sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188
Authorization: Digest username=“19995551313”,realm=“asterisk”,cnonce=“327b23c6”,nc=00000001,qop=auth,uri=“sip:192.22.33.188:5060”,nonce=“1523032815/fe49c999a305e631534ab1dbbb95f8bd”,response=“e6ec434c8f4e7230070456718082d29b”,algorithm=md5,opaque=“164da8996bc530ae”
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
<— Transmitting SIP request (986 bytes) to UDP:192.22.33.214:5060 —>
INVITE sip:19995551212@192.22.33.214:5060 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPjbfb4f7ea-74aa-4673-9504-715b3b98a17b
From: sip:19995551313@192.22.33.188;tag=8ef9e442-bb96-4014-88b6-932f9eb8bec0
To: sip:19995551212@192.22.33.214
Contact: sip:asterisk@192.22.33.188:5060
Call-ID: 30da62c5-00e5-444c-8fdf-1f8a625cdd07
CSeq: 17234 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 15.1.2
Content-Type: application/sdp
Content-Length: 310
v=0
o=- 246335027 246335027 IN IP4 192.22.33.188
s=Asterisk
c=IN IP4 192.22.33.188
t=0 0
m=audio 47690 RTP/AVP 0 8 3 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Received SIP response (390 bytes) from UDP:192.22.33.214:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPjbfb4f7ea-74aa-4673-9504-715b3b98a17b
From: sip:19995551313@192.22.33.188;tag=8ef9e442-bb96-4014-88b6-932f9eb8bec0
To: sip:19995551212@192.22.33.214;tag=184SIPpTag0143
Call-ID: 30da62c5-00e5-444c-8fdf-1f8a625cdd07
CSeq: 17234 INVITE
Contact: sip:192.22.33.214:5060;transport=UDP
Content-Length: 0
<— Received SIP response (801 bytes) from UDP:192.22.33.214:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPjbfb4f7ea-74aa-4673-9504-715b3b98a17b
From: sip:19995551313@192.22.33.188;tag=8ef9e442-bb96-4014-88b6-932f9eb8bec0
To: sip:19995551212@192.22.33.214;tag=184SIPpTag0143
Call-ID: 30da62c5-00e5-444c-8fdf-1f8a625cdd07
CSeq: 17234 INVITE
Server: DID Logic GW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Contact: sip:192.22.33.214:5060;transport=UDP
Content-Type: application/sdp
Content-Length: 276
v=0
o=user1 53655765 2353687637 IN IP4 192.22.33.214
s=DID Logic MGW
c=IN IP4 192.38.11.230
t=0 0
m=audio 11512 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<— Transmitting SIP request (419 bytes) to UDP:192.22.33.214:5060 —>
ACK sip:192.22.33.214:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPj9ea4b810-c7fc-4b83-85da-5e03462ac259
From: sip:19995551313@192.22.33.188;tag=8ef9e442-bb96-4014-88b6-932f9eb8bec0
To: sip:19995551212@192.22.33.214;tag=184SIPpTag0143
Call-ID: 30da62c5-00e5-444c-8fdf-1f8a625cdd07
CSeq: 17234 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 15.1.2
Content-Length: 0
[Apr 6 16:40:51] DEBUG[31832][C-00000002]: channel.c:3192 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Apr 6 16:40:52] DEBUG[31832][C-00000002]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:52] DEBUG[31832][C-00000002]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:52] DEBUG[31832][C-00000002]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:52] DEBUG[31832][C-00000002]: channel.c:5551 set_format: Channel PJSIP/19995551313-00000001 setting write format path: ulaw -> ulaw
[Apr 6 16:40:55] DEBUG[31844]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:55] DEBUG[31844]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:55] DEBUG[31844]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:55] DEBUG[31844]: channel.c:5551 set_format: Channel PJSIP/192.22.33.214-00000002 setting write format path: ulaw -> ulaw
[Apr 6 16:40:55] DEBUG[31863]: http.c:1914 httpd_helper_thread: HTTP opening session. Top level
[Apr 6 16:40:55] DEBUG[31863]: http.c:1390 handle_uri: HTTP Request URI is /ari/channels/1523032827.2/play?api_key=ari4java:1234&media=sound%3A%2Fvar%2Flib%2Fasterisk%2Fsounds%2Fen%2FPrivacyStar%2FSCR_caller_connect&offsetms=0&skipms=3000
[Apr 6 16:40:55] DEBUG[31863]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/play] with handler [httpstatus] len 10
[Apr 6 16:40:55] DEBUG[31863]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/play] with handler [static] len 6
[Apr 6 16:40:55] DEBUG[31863]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/play] with handler [ari] len 3
[Apr 6 16:40:55] DEBUG[31863]: http.c:1461 handle_uri: Match made with [ari]
[Apr 6 16:40:55] DEBUG[31863]: http.c:551 ast_http_send: HTTP keeping session open. status_code:201
[Apr 6 16:40:55] DEBUG[31844]: media_cache.c:250 ast_media_cache_retrieve: Failed to obtain media at ‘/var/lib/asterisk/sounds/en/PrivacyStar/SCR_caller_connect’
[Apr 6 16:40:55] DEBUG[31844]: media_cache.c:250 ast_media_cache_retrieve: Failed to obtain media at ‘/var/lib/asterisk/sounds/en/PrivacyStar/SCR_caller_connect’
[Apr 6 16:40:55] DEBUG[31844]: channel.c:5551 set_format: Channel PJSIP/192.22.33.214-00000002 setting write format path: gsm -> ulaw
[Apr 6 16:40:55] DEBUG[31844]: channel.c:3192 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Apr 6 16:40:55] DEBUG[31863]: http.c:1971 httpd_helper_thread: HTTP closing session. Top level
[Apr 6 16:40:56] DEBUG[31844]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:56] DEBUG[31844]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:56] DEBUG[31844]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:40:56] DEBUG[31844]: channel.c:5551 set_format: Channel PJSIP/192.22.33.214-00000002 setting write format path: ulaw -> ulaw
[Apr 6 16:40:57] DEBUG[31866]: http.c:1914 httpd_helper_thread: HTTP opening session. Top level
[Apr 6 16:40:57] DEBUG[31866]: http.c:1390 handle_uri: HTTP Request URI is /ari/channels/1523032827.2/variable?api_key=ari4java:1234&variable=CHANNEL%28pjsip%2Ccall-id%29
[Apr 6 16:40:57] DEBUG[31866]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [httpstatus] len 10
[Apr 6 16:40:57] DEBUG[31866]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [static] len 6
[Apr 6 16:40:57] DEBUG[31866]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [ari] len 3
[Apr 6 16:40:57] DEBUG[31866]: http.c:1461 handle_uri: Match made with [ari]
[Apr 6 16:40:57] DEBUG[31866]: http.c:551 ast_http_send: HTTP keeping session open. status_code:200
[Apr 6 16:40:57] DEBUG[31866]: http.c:1971 httpd_helper_thread: HTTP closing session. Top level
[Apr 6 16:40:57] DEBUG[31867]: http.c:1914 httpd_helper_thread: HTTP opening session. Top level
[Apr 6 16:40:57] DEBUG[31867]: http.c:1390 handle_uri: HTTP Request URI is /ari/channels/1523032827.2/variable?api_key=ari4java:1234&variable=CHANNEL%28pjsip%2Clocal_tag%29
[Apr 6 16:40:57] DEBUG[31867]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [httpstatus] len 10
[Apr 6 16:40:57] DEBUG[31867]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [static] len 6
[Apr 6 16:40:57] DEBUG[31867]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [ari] len 3
[Apr 6 16:40:57] DEBUG[31867]: http.c:1461 handle_uri: Match made with [ari]
[Apr 6 16:40:57] DEBUG[31867]: http.c:551 ast_http_send: HTTP keeping session open. status_code:200
[Apr 6 16:40:57] DEBUG[31867]: http.c:1971 httpd_helper_thread: HTTP closing session. Top level
[Apr 6 16:40:57] DEBUG[31868]: http.c:1914 httpd_helper_thread: HTTP opening session. Top level
[Apr 6 16:40:57] DEBUG[31868]: http.c:1390 handle_uri: HTTP Request URI is /ari/channels/1523032827.2/variable?api_key=ari4java:1234&variable=CHANNEL%28pjsip%2Cremote_tag%29
[Apr 6 16:40:57] DEBUG[31868]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [httpstatus] len 10
[Apr 6 16:40:57] DEBUG[31868]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [static] len 6
[Apr 6 16:40:57] DEBUG[31868]: http.c:1445 handle_uri: match request [ari/channels/1523032827.2/variable] with handler [ari] len 3
[Apr 6 16:40:57] DEBUG[31868]: http.c:1461 handle_uri: Match made with [ari]
[Apr 6 16:40:57] DEBUG[31868]: http.c:551 ast_http_send: HTTP keeping session open. status_code:200
[Apr 6 16:40:58] DEBUG[31868]: http.c:1971 httpd_helper_thread: HTTP closing session. Top level
[Apr 6 16:40:58] DEBUG[31869]: http.c:1914 httpd_helper_thread: HTTP opening session. Top level
[Apr 6 16:40:58] DEBUG[31869]: http.c:1390 handle_uri: HTTP Request URI is /ari/channels/1523032815.1/redirect?api_key=ari4java:1234&endpoint=PJSIP%2Fsip%3A19995551212%40192.22.33.214%3FReplaces%3D30da62c5-00e5-444c-8fdf-1f8a625cdd07%253Bto-tag%253D184SIPpTag0143%253Bfrom-tag%253D8ef9e442-bb96-4014-88b6-932f9eb8bec0
[Apr 6 16:40:58] DEBUG[31869]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1/redirect] with handler [httpstatus] len 10
[Apr 6 16:40:58] DEBUG[31869]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1/redirect] with handler [static] len 6
[Apr 6 16:40:58] DEBUG[31869]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1/redirect] with handler [ari] len 3
[Apr 6 16:40:58] DEBUG[31869]: http.c:1461 handle_uri: Match made with [ari]
[Apr 6 16:40:58] DEBUG[31869]: http.c:551 ast_http_send: HTTP keeping session open. status_code:204
[Apr 6 16:40:58] DEBUG[31740]: res_pjsip/pjsip_resolver.c:477 sip_resolve: Performing SIP DNS resolution of target ‘192.22.33.42’
[Apr 6 16:40:58] DEBUG[31740]: res_pjsip/pjsip_resolver.c:504 sip_resolve: Transport type for target ‘192.22.33.42’ is ‘UDP’
[Apr 6 16:40:58] DEBUG[31740]: res_pjsip/pjsip_resolver.c:525 sip_resolve: Target ‘192.22.33.42’ is an IP address, skipping resolution
<— Transmitting SIP request (853 bytes) to UDP:192.22.33.42:5060 —>
REFER sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPj6ea9cf0f-1649-4cfd-9037-e7c4560ffcdc
From: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
To: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
Contact: sip:192.22.33.188:5060
Call-ID: 1-650@192.17.0.2
CSeq: 5948 REFER
Event: refer
Expires: 600
Supported: 100rel, timer, replaces, norefersub
Accept: message/sipfrag;version=2.0
Allow-Events: presence, dialog, message-summary, refer
Refer-To: sip:19995551212@192.22.33.214?Replaces=30da62c5-00e5-444c-8fdf-1f8a625cdd07%3Bto-tag%3D184SIPpTag0143%3Bfrom-tag%3D8ef9e442-bb96-4014-88b6-932f9eb8bec0
Referred-By: sip:19995551212@192.22.33.188
Max-Forwards: 70
User-Agent: Asterisk PBX 15.1.2
Content-Length: 0
<— Received SIP response (493 bytes) from UDP:192.22.33.42:5060 —>
SIP/2.0 202 Accepted
CSeq: 5948 REFER
Call-ID: 1-650@192.17.0.2
From: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
To: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPj6ea9cf0f-1649-4cfd-9037-e7c4560ffcdc;received=192.22.33.188
Contact: “SIPpTest” sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188
Subject: Performance Test
Content-Length: 0
[Apr 6 16:40:58] DEBUG[31740]: res_pjsip_session.c:3565 session_inv_on_tsx_state_changed: REFER received final response code 202
<— Received SIP request (906 bytes) from UDP:192.22.33.42:5060 —>
NOTIFY sip:192.22.33.188:5060 SIP/2.0
CSeq: 5949 NOTIFY
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
Call-ID: 1-650@192.17.0.2
Max-Forwards: 70
Via: SIP/2.0/UDP 192.22.33.42:5060;branch=z9hG4bKPj6ea9cf0f-1649-4cfd-9037-e7c4560ffcdc
Contact: “SIPpTest” sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Authorization: Digest username=“19995551313”,realm=“asterisk”,cnonce=“643c9869”,nc=00000001,qop=auth,uri=“sip:192.22.33.188:5060”,nonce=“1523032815/fe49c999a305e631534ab1dbbb95f8bd”,response=“3825a6c9890f85ce911f946e9053fc06”,algorithm=md5,opaque=“164da8996bc530ae”
Event: refer
Subscription-State: active;expires=60
Content-Type: message/sipfrag
Subject: Performance Test
Content-Length: 20
SIP/2.0 100 Trying
<— Transmitting SIP response (562 bytes) to UDP:192.22.33.42:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.22.33.42:5060;rport=5060;received=192.22.33.42;branch=z9hG4bKPj6ea9cf0f-1649-4cfd-9037-e7c4560ffcdc
Call-ID: 1-650@192.17.0.2
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
CSeq: 5949 NOTIFY
Contact: sip:192.22.33.188:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 15.1.2
Content-Length: 0
[Apr 6 16:40:58] DEBUG[31869]: http.c:1971 httpd_helper_thread: HTTP closing session. Top level
[Apr 6 16:41:04] DEBUG[31799]: db.c:429 ast_db_del: Unable to find key ‘6001;@0236855fa4aec8cf6581e6b184d7b6c9’ in family ‘registrar/contact’
<— Received SIP request (371 bytes) from UDP:192.22.33.214:5060 —>
BYE sip:asterisk@192.22.33.188:5060 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.214:5060
From: sip:19995551212@192.22.33.214;tag=184SIPpTag0143
To: sip:19995551313@192.22.33.188;tag=8ef9e442-bb96-4014-88b6-932f9eb8bec0
Call-ID: 30da62c5-00e5-444c-8fdf-1f8a625cdd07
Cseq: 1 BYE
Contact: sip:sipp@192.22.33.214:5060;transport=UDP
Max-Forwards: 70
Content-Length: 0
<— Transmitting SIP response (334 bytes) to UDP:192.22.33.214:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.22.33.214:5060;rport=5060;received=192.22.33.214
Call-ID: 30da62c5-00e5-444c-8fdf-1f8a625cdd07
From: sip:19995551212@192.22.33.214;tag=184SIPpTag0143
To: sip:19995551313@192.22.33.188;tag=8ef9e442-bb96-4014-88b6-932f9eb8bec0
CSeq: 1 BYE
Server: Asterisk PBX 15.1.2
Content-Length: 0
[Apr 6 16:41:13] DEBUG[31844]: channel.c:2587 ast_hangup: Channel 0x7f08801eef98 ‘PJSIP/192.22.33.214-00000002’ hanging up. Refs: 2
[Apr 6 16:41:13] DEBUG[31844]: chan_pjsip.c:2251 hangup_cause2sip: AST hangup cause 16 (no match found in PJSIP)
[Apr 6 16:41:13] DEBUG[31740]: rtp_engine.c:414 instance_destructor: Destroyed RTP instance ‘0x7f087800b758’
[Apr 6 16:41:13] DEBUG[31740]: channel.c:2222 ast_channel_destructor: Channel 0x7f08801eef98 ‘PJSIP/192.22.33.214-00000002’ destroying
[Apr 6 16:41:13] DEBUG[31732]: cdr.c:1289 cdr_object_finalize: Finalized CDR for PJSIP/192.22.33.214-00000002 - start 1523032827.246045 answer 1523032827.249382 end 1523032873.257398 dispo ANSWERED
[Apr 6 16:41:13] DEBUG[31878]: http.c:1914 httpd_helper_thread: HTTP opening session. Top level
[Apr 6 16:41:13] DEBUG[31878]: http.c:1390 handle_uri: HTTP Request URI is /ari/channels/1523032815.1/play?api_key=ari4java:1234&media=sound%3A%2Fvar%2Flib%2Fasterisk%2Fsounds%2Fen%2FPrivacyStar%2FSCR_caller_calledhangup&offsetms=0&skipms=3000
[Apr 6 16:41:13] DEBUG[31878]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1/play] with handler [httpstatus] len 10
[Apr 6 16:41:13] DEBUG[31878]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1/play] with handler [static] len 6
[Apr 6 16:41:13] DEBUG[31878]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1/play] with handler [ari] len 3
[Apr 6 16:41:13] DEBUG[31878]: http.c:1461 handle_uri: Match made with [ari]
[Apr 6 16:41:13] DEBUG[31878]: http.c:551 ast_http_send: HTTP keeping session open. status_code:201
[Apr 6 16:41:13] DEBUG[31832][C-00000002]: media_cache.c:250 ast_media_cache_retrieve: Failed to obtain media at ‘/var/lib/asterisk/sounds/en/PrivacyStar/SCR_caller_calledhangup’
[Apr 6 16:41:13] DEBUG[31832][C-00000002]: media_cache.c:250 ast_media_cache_retrieve: Failed to obtain media at ‘/var/lib/asterisk/sounds/en/PrivacyStar/SCR_caller_calledhangup’
[Apr 6 16:41:13] DEBUG[31832][C-00000002]: channel.c:5551 set_format: Channel PJSIP/19995551313-00000001 setting write format path: gsm -> ulaw
[Apr 6 16:41:13] DEBUG[31832][C-00000002]: channel.c:3192 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Apr 6 16:41:13] DEBUG[31878]: http.c:1971 httpd_helper_thread: HTTP closing session. Top level
<— Received SIP request (646 bytes) from UDP:192.22.33.42:5060 —>
NOTIFY sip:192.22.33.188:5060 SIP/2.0
CSeq: 5950 NOTIFY
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
Call-ID: 1-650@192.17.0.2
Max-Forwards: 70
Via: SIP/2.0/UDP 192.22.33.42:5060;branch=z9hG4bKPj6ea9cf0f-1649-4cfd-9037-e7c4560ffcdc
Contact: “SIPpTest” sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Event: refer
Subscription-State: terminated;reason=noresource
Content-Type: message/sipfrag
Subject: Performance Test
Content-Length: 16
SIP/2.0 200 OK
<— Transmitting SIP response (562 bytes) to UDP:192.22.33.42:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.22.33.42:5060;rport=5060;received=192.22.33.42;branch=z9hG4bKPj6ea9cf0f-1649-4cfd-9037-e7c4560ffcdc
Call-ID: 1-650@192.17.0.2
From: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
To: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
CSeq: 5949 NOTIFY
Contact: sip:192.22.33.188:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 15.1.2
Content-Length: 0
[Apr 6 16:41:18] DEBUG[31832][C-00000002]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:41:18] DEBUG[31832][C-00000002]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:41:18] DEBUG[31832][C-00000002]: channel.c:3192 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Apr 6 16:41:18] DEBUG[31832][C-00000002]: channel.c:5551 set_format: Channel PJSIP/19995551313-00000001 setting write format path: ulaw -> ulaw
[Apr 6 16:41:18] DEBUG[31881]: http.c:1914 httpd_helper_thread: HTTP opening session. Top level
[Apr 6 16:41:18] DEBUG[31881]: http.c:1390 handle_uri: HTTP Request URI is /ari/channels/1523032815.1?api_key=ari4java:1234
[Apr 6 16:41:18] DEBUG[31881]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1] with handler [httpstatus] len 10
[Apr 6 16:41:18] DEBUG[31881]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1] with handler [static] len 6
[Apr 6 16:41:18] DEBUG[31881]: http.c:1445 handle_uri: match request [ari/channels/1523032815.1] with handler [ari] len 3
[Apr 6 16:41:18] DEBUG[31881]: http.c:1461 handle_uri: Match made with [ari]
[Apr 6 16:41:18] DEBUG[31881]: channel.c:2496 ast_softhangup_nolock: Soft-Hanging (0x20) up channel ‘PJSIP/19995551313-00000001’
[Apr 6 16:41:18] DEBUG[31832][C-00000002]: pbx.c:4365 __ast_pbx_run: Extension 19995551212, priority 3 returned normally even though call was hung up
[Apr 6 16:41:18] DEBUG[31832][C-00000002]: channel.c:2496 ast_softhangup_nolock: Soft-Hanging (0x10) up channel ‘PJSIP/19995551313-00000001’
[Apr 6 16:41:18] DEBUG[31832][C-00000002]: channel.c:2587 ast_hangup: Channel 0x7f0878021ad8 ‘PJSIP/19995551313-00000001’ hanging up. Refs: 3
[Apr 6 16:41:18] DEBUG[31832][C-00000002]: chan_pjsip.c:2251 hangup_cause2sip: AST hangup cause 16 (no match found in PJSIP)
[Apr 6 16:41:18] DEBUG[31740]: rtp_engine.c:414 instance_destructor: Destroyed RTP instance ‘0x7f087800bcd8’
[Apr 6 16:41:18] DEBUG[31740]: res_pjsip/pjsip_resolver.c:477 sip_resolve: Performing SIP DNS resolution of target ‘192.22.33.42’
[Apr 6 16:41:18] DEBUG[31740]: res_pjsip/pjsip_resolver.c:504 sip_resolve: Transport type for target ‘192.22.33.42’ is ‘UDP’
[Apr 6 16:41:18] DEBUG[31740]: res_pjsip/pjsip_resolver.c:525 sip_resolve: Target ‘192.22.33.42’ is an IP address, skipping resolution
<— Transmitting SIP request (460 bytes) to UDP:192.22.33.42:5060 —>
BYE sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPj2024d904-a0ce-473f-afe1-fe34980486d5
From: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
To: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
Call-ID: 1-650@192.17.0.2
CSeq: 5949 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 15.1.2
Content-Length: 0
[Apr 6 16:41:18] DEBUG[31881]: channel.c:2222 ast_channel_destructor: Channel 0x7f0878021ad8 ‘PJSIP/19995551313-00000001’ destroying
[Apr 6 16:41:18] DEBUG[31732]: cdr.c:1289 cdr_object_finalize: Finalized CDR for PJSIP/19995551313-00000001 - start 1523032815.807337 answer 1523032815.807631 end 1523032878.625770 dispo ANSWERED
[Apr 6 16:41:18] DEBUG[31881]: http.c:551 ast_http_send: HTTP keeping session open. status_code:204
[Apr 6 16:41:18] DEBUG[31881]: http.c:1971 httpd_helper_thread: HTTP closing session. Top level
<— Transmitting SIP request (460 bytes) to UDP:192.22.33.42:5060 —>
BYE sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPj2024d904-a0ce-473f-afe1-fe34980486d5
From: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
To: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
Call-ID: 1-650@192.17.0.2
CSeq: 5949 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 15.1.2
Content-Length: 0
<— Received SIP response (357 bytes) from UDP:192.22.33.42:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPj2024d904-a0ce-473f-afe1-fe34980486d5
From: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
To: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
Call-ID: 1-650@192.17.0.2
CSeq: 5949 BYE
Contact: sip:192.22.33.42:5060;transport=UDP
Content-Length: 0
[Apr 6 16:41:19] DEBUG[31740]: res_pjsip_session.c:3565 session_inv_on_tsx_state_changed: BYE received final response code 200
[Apr 6 16:41:34] DEBUG[31799]: db.c:429 ast_db_del: Unable to find key ‘6001;@0236855fa4aec8cf6581e6b184d7b6c9’ in family ‘registrar/contact’
[Apr 6 16:41:38] DEBUG[31833]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying.
[Apr 6 16:41:53] DEBUG[31790]: res_pjsip/pjsip_resolver.c:477 sip_resolve: Performing SIP DNS resolution of target ‘192.22.33.42’
[Apr 6 16:41:53] DEBUG[31790]: res_pjsip/pjsip_resolver.c:504 sip_resolve: Transport type for target ‘192.22.33.42’ is ‘UDP’
[Apr 6 16:41:53] DEBUG[31790]: res_pjsip/pjsip_resolver.c:525 sip_resolve: Target ‘192.22.33.42’ is an IP address, skipping resolution
<— Transmitting SIP request (652 bytes) to UDP:192.22.33.42:5060 —>
SUBSCRIBE sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPj8652030f-9438-4fdb-8b77-565e08efefd4
From: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
To: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
Contact: sip:192.22.33.188:5060
Call-ID: 1-650@192.17.0.2
CSeq: 5950 SUBSCRIBE
Event: refer
Expires: 600
Supported: 100rel, timer, replaces, norefersub
Accept: message/sipfrag;version=2.0
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 15.1.2
Content-Length: 0
<— Transmitting SIP request (652 bytes) to UDP:192.22.33.42:5060 —>
SUBSCRIBE sip:19995551313@192.22.33.42:5060;transport=udp;registering_acc=192_22_33_188 SIP/2.0
Via: SIP/2.0/UDP 192.22.33.188:5060;rport;branch=z9hG4bKPj8652030f-9438-4fdb-8b77-565e08efefd4
From: sip:19995551212@192.22.33.188;tag=fa6b1828-e64b-41c2-8f71-6ae090ab309d
To: “SIPpTest” sip:19995551313@192.22.33.42;tag=1
Contact: sip:192.22.33.188:5060
Call-ID: 1-650@192.17.0.2
CSeq: 5950 SUBSCRIBE
Event: refer
Expires: 600
Supported: 100rel, timer, replaces, norefersub
Accept: message/sipfrag;version=2.0
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 15.1.2
Content-Length: 0
…more identical SUBSCRIBES but I need to cut down on my character count…