Asterisk does not send BYE to transferer after it gets REFER for blind transfer

Asterisk Asterisk SIP
1

Hello, I have a blind transfer call scenario where A-party calls B-party and then B party sends blind transfer REFER to refer A party to C party. In general the transfer works fine, the only problem I have is that I would like Asterisk to send BYE to transferer B-party after it confirms REFER by 202 or 200 message but always when bridge to the B party is destroyed when the transfer takes place, I can see the following in debug log:

[Jan 10 14:22:22] DEBUG[183371][C-0000027e] channel.c: Channel 0x7f3660244700 ‘PJSIP/9833-0000056d’ hanging up. Refs: 2
[Jan 10 14:22:22] DEBUG[183371][C-0000027e] chan_pjsip.c: PJSIP/9833-0000056d
[Jan 10 14:22:22] DEBUG[183371][C-0000027e] chan_pjsip.c: AST hangup cause 16 (no match found in PJSIP)
[Jan 10 14:22:22] DEBUG[183366] chan_pjsip.c: PJSIP/9833-0000056d
[Jan 10 14:22:22] DEBUG[183371][C-0000027e] chan_pjsip.c: Cause: 0
[Jan 10 14:22:22] DEBUG[183366] res_pjsip_session.c: PJSIP/9833-0000056d Response 0
[Jan 10 14:22:22] DEBUG[183366] res_pjsip_session.c: Deferred
[Jan 10 14:22:22] DEBUG[183366] chan_pjsip.c:

The “PJSIP/9833-0000056d” is the transferer B-party channel and instead of sending BYE there is this log “Deferred”. If I compare this log snippet with a call release of a party in a simple call scenario A calls B, the footprint is exactly the same only instead of the event “Deferred” the BYE is sent out.

So far I tried the following things but without success:

  • I could not see in debug log that Asterisk would start Hangup app for the transferer B party channel so I split the diaplans for calling and called party by Dial() G option, so before the transfer when A party calls B party the Hangup can be started for each channel independently. This worked as I could see “Launching ‘Hangup’” in debug for the transferer B-party channel but still the BYE was not sent and was “Deferred” instead.

  • In res-pjsip-refer.c code comments I noticed note that after the transfer the transferer channel is locked so Asterisk can send NOTIFY messages to the transferer about the progress of the transfer, thus I assumed that if B-party does not subscribe to transfer progress NOTIFYs by adding Refer-Sub:false header to REFER message, the Asterisk would be able to send the BYE but this did not work either, the result was still the same.

Can you pls help me to figure out why there is this “Deferred” event and how to send the BYE together with “Soft-Hanging” the transferer channel?

Br

Zdenek

2

Howdy!

Interesting find. Digging in to “norefersub” a bit, and associated global setting for pjsip.conf, and it appears broken in current chan_pjsip, having no effect when toggled on or off, thus (potentially) taking Asterisk out of compliance with RFC4488 (but maybe it is respecting Refer-Sub so partially correct ?)

What does your SIP trace look like from the B-party device ? eg. “pjsip set logger host 1.2.3.4”

3

Hello penguinpbx, thanks a lot for your reply and sorry for my delayed response. The attached pjsip_log_3 contains the packets of the whole call. The flow is as follows:

  • A:+420112855219 calls B:+420112809833 (asterisk extension 9833)
  • Some time after B answers the call, it sends blind transfer REFER, referring to C:+420115712700
  • Asterisk correctly sends INVITEs to A and C and after C accepts the call A and C can hear each other.
  • Then A hangs up the call.

When REFER contains the “Refer-Sub: false” header, asterisk correctly unsbsribes the B from transfer progress NOTIFYs, also I can see in debug log the following message which is correct for this situation I believe:

[Jan 26 12:51:47] DEBUG[844947] res_pjsip_refer.c: Progress monitoring not requested for REFER on channel ‘PJSIP/9833-0000020f’ from endpoint ‘9833’, sending immediate response of ‘200’

Thus IMHO the Refer-Sub header is correctly reflected but what I mainly have problem with is that Asterisk never sends BYE to the B party after it Hangs up the B party channel. In debug I see following when B party channel is hanged:

[Jan 26 12:51:47] DEBUG[844985] channel.c: Soft-Hanging (0x20) up channel ‘PJSIP/9833-0000020f’
[Jan 26 12:51:47] VERBOSE[844985] asterisk.c: Remote UNIX connection disconnected
[Jan 26 12:51:47] DEBUG[844973][C-000000ed] channel.c: Channel 0x7fa5d80a1440 ‘PJSIP/9833-0000020f’ hanging up. Refs: 2
[Jan 26 12:51:47] DEBUG[844973][C-000000ed] chan_pjsip.c: PJSIP/9833-0000020f
[Jan 26 12:51:47] DEBUG[844973][C-000000ed] chan_pjsip.c: AST hangup cause 16 (no match found in PJSIP)
[Jan 26 12:51:47] DEBUG[844973][C-000000ed] chan_pjsip.c: Cause: 0
[Jan 26 12:51:47] DEBUG[844947] chan_pjsip.c: PJSIP/9833-0000020f
[Jan 26 12:51:47] DEBUG[844947] res_pjsip_session.c: PJSIP/9833-0000020f Response 0
[Jan 26 12:51:47] DEBUG[844947] res_pjsip_session.c: Deferred

Instead of sending BYE, there is this “Deferred” event, do you have any idea what it means?

I also tried to explicitly hangup the B (called) party by hanging it up from dialplan in extensions.conf by the following line:

exten => h,1,System(asterisk -rx ‘channel request hangup ${CHANNEL}’)

But it also does not have any effect, this is the snippet from the debug:

[Jan 26 12:51:47] VERBOSE[844973][C-000000ed] pbx.c: Auto fallthrough, channel ‘PJSIP/9833-0000020f’ status is ‘UNKNOWN’
[Jan 26 12:51:47] DEBUG[844973][C-000000ed] channel.c: Soft-Hanging (0x10) up channel ‘PJSIP/9833-0000020f’
[Jan 26 12:51:47] DEBUG[844973][C-000000ed] channel.c: Soft-Hanging (0x80) up channel ‘PJSIP/9833-0000020f’
[Jan 26 12:51:47] DEBUG[844973][C-000000ed] pbx_variables.c: Result of ‘CHANNEL’ is ‘PJSIP/9833-0000020f’
[Jan 26 12:51:47] DEBUG[844973][C-000000ed] pbx.c: Launching ‘System’
[Jan 26 12:51:47] VERBOSE[844973][C-000000ed] pbx.c: Executing [h@called:1] System(“PJSIP/9833-0000020f”, “asterisk -rx ‘channel request hangup PJSIP/9833-0000020f’”) in new stack

and nothing more related to the channel PJSIP/9833-0000020f.

Originally I thought asterisk waits until the C party (transfer target) picks up the call and sends 200OK but in the attached packet trace you can see that even after C party picking up, the BYE is never sent from asterisk towards B party.

Asterisk keeps the session with B party open even after A and C exchange BYE at the end of the call, only later B party sends BYE when it times out.

One thing I was thinking about is that there is an issue with B party’s UA that it sends REFER and BYE with incorrect sequence in Cseq, can the “Deferred” message in debug come from the incorrect CSeq sequence ?

4

sorry I just found out I can’t add attachments as new user so adding it here. I tried to reduce it a bit by removing SDPs

[Jan 26 11:53:05] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP request (1712 bytes) from TCP:192.168.129.85:18116 --->
INVITE sip:+420112809833@10.10.54.20:5060;transport=tcp;line=gejcclb SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/TCP 192.168.129.85:5060;branch=z9hG4bKg3Zqkv7ivnoljf39uviip8ew248xlxqwm
To: <sip:+420112809833@domain1;user=phone>;cscf
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 583992421 INVITE
Contact: <sip:sgc_c@192.168.129.85;transport=tcp>
Record-Route: <sip:192.168.129.85;transport=tcp;lr>
Accept-Contact: *;explicit;description="<sip:+420119001486@domain1>";require
Min-Se: 900
P-Asserted-Identity: <sip:+420112855219@domain1>
P-Asserted-Identity: <tel:+420112855219>
P-Early-Media: supported
Privacy: none
Session-Expires: 1800;refresher=uac
Supported: 100rel
Supported: timer
Content-Type: application/sdp
Content-Length: 188
Recv-Info: x-broadworks-client-session-info
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY

[Jan 26 11:53:05] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (495 bytes) to TCP:192.168.129.85:18116 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.129.85:5060;rport=18116;received=192.168.129.85;branch=z9hG4bKg3Zqkv7ivnoljf39uviip8ew248xlxqwm
Record-Route: <sip:192.168.129.85:18116;transport=TCP;lr>
Call-ID: BW1253213522601241916308192@10.2.137.167
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
To: <sip:+420112809833@domain1;user=phone>;cscf
CSeq: 583992421 INVITE
Server: asterisk1
Content-Length:  0


[Jan 26 11:53:05] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (923 bytes) to UDP:10.10.54.45:5241 --->
INVITE sip:9833@10.10.54.45:5241;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.54.20:5060;branch=z9hG4bKPjcdaa60fc-c717-4cec-a1bf-5b953307dc81
From: <sip:+420112855219@10.113.157.108>;tag=02460ee6-0800-408d-a406-6b9ba32d7780
To: <sip:9833@10.10.54.45>
Contact: <sip:asterisk@10.10.54.20:5060>
Call-ID: 198abaf5-9dc7-4515-919d-6eef3c64a01b
CSeq: 30878 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: asterisk1
Content-Type: application/sdp
Content-Length:   254

[Jan 26 11:53:05] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (394 bytes) from UDP:10.10.54.45:5241 --->
SIP/2.0 180 Ringing
CSeq: 30878 INVITE
Call-ID: 198abaf5-9dc7-4515-919d-6eef3c64a01b
From: <sip:+420112855219@10.113.157.108>;tag=02460ee6-0800-408d-a406-6b9ba32d7780
To: <sip:9833@10.10.54.45>;tag=Agent-2
Via: SIP/2.0/UDP 10.10.54.20:5060;branch=z9hG4bKPjcdaa60fc-c717-4cec-a1bf-5b953307dc81
Contact: <sip:9833@10.10.54.45:5241;transport=udp>;expires=479
Content-Length: 0


[Jan 26 11:53:05] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (724 bytes) to TCP:192.168.129.85:18116 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.129.85:5060;rport=18116;received=192.168.129.85;branch=z9hG4bKg3Zqkv7ivnoljf39uviip8ew248xlxqwm
Record-Route: <sip:192.168.129.85:18116;transport=TCP;lr>
Call-ID: BW1253213522601241916308192@10.2.137.167
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
To: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
CSeq: 583992421 INVITE
Server: asterisk1
Contact: <sip:10.10.54.20:5060;transport=TCP>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Require: 100rel
RSeq: 10975
Content-Length:  0


[Jan 26 11:53:06] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP request (489 bytes) from TCP:192.168.129.85:18116 --->
PRACK sip:10.10.54.20:5060;transport=tcp SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/TCP 192.168.129.85:5060;branch=z9hG4bKg3Zqkv7ikyscb9be83nllkuoo4jryeqfm
To: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 583992422 PRACK
RAck: 10975 583992421 INVITE
Content-Length: 0


[Jan 26 11:53:06] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (472 bytes) to TCP:192.168.129.85:18116 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.129.85:5060;rport=18116;received=192.168.129.85;branch=z9hG4bKg3Zqkv7ikyscb9be83nllkuoo4jryeqfm
Call-ID: BW1253213522601241916308192@10.2.137.167
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
To: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
CSeq: 583992422 PRACK
Server: asterisk1
Content-Length:  0


[Jan 26 11:53:06] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (592 bytes) from UDP:10.10.54.45:5241 --->
SIP/2.0 200 OK
CSeq: 30878 INVITE
Call-ID: 198abaf5-9dc7-4515-919d-6eef3c64a01b
From: <sip:+420112855219@10.113.157.108>;tag=02460ee6-0800-408d-a406-6b9ba32d7780
To: <sip:9833@10.10.54.45>;tag=Agent-2
Via: SIP/2.0/UDP 10.10.54.20:5060;branch=z9hG4bKPjcdaa60fc-c717-4cec-a1bf-5b953307dc81
Content-Type: application/sdp
Contact: <sip:9833@10.10.54.45:5241;transport=udp>;expires=479
Content-Length: 170

v=0
o=Telekom-SDP 1706270001399 1 IN IP4 10.10.54.45
s=Telekom-SDP
c=IN IP4 10.10.54.45
t=0 0
m=audio 17104 RTP/AVP 9
a=rtpmap:9 G722/8000/1
a=ptime:20
a=sendrecv

[Jan 26 11:53:06] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (409 bytes) to UDP:10.10.54.45:5241 --->
ACK sip:9833@10.10.54.45:5241;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.54.20:5060;branch=z9hG4bKPj690a2bd3-42a6-4c35-9a57-72b4e310d0ee
From: <sip:+420112855219@10.113.157.108>;tag=02460ee6-0800-408d-a406-6b9ba32d7780
To: <sip:9833@10.10.54.45>;tag=Agent-2
Call-ID: 198abaf5-9dc7-4515-919d-6eef3c64a01b
CSeq: 30878 ACK
Max-Forwards: 70
User-Agent: asterisk1
Content-Length:  0


[Jan 26 11:53:06] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (989 bytes) to TCP:192.168.129.85:18116 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.129.85:5060;rport=18116;received=192.168.129.85;branch=z9hG4bKg3Zqkv7ivnoljf39uviip8ew248xlxqwm
Record-Route: <sip:192.168.129.85:18116;transport=TCP;lr>
Call-ID: BW1253213522601241916308192@10.2.137.167
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
To: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
CSeq: 583992421 INVITE
Server: asterisk1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.10.54.20:5060;transport=TCP>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   165

[Jan 26 11:53:06] WARNING[843179][C-000000ec] app_bridgewait.c: Failed to join waiting bridge 'default' for 'Surrogate/PJSIP/Trunk_1a_in-0000020b'.
[Jan 26 11:53:06] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP request (506 bytes) from TCP:192.168.129.85:18116 --->
ACK sip:10.10.54.20:5060;transport=tcp SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/TCP 192.168.129.85:5060;branch=z9hG4bKg3Zqkv7ijp3j4ku1tm1hb763p8lmro62s
To: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 583992421 ACK
Contact: <sip:sgc_c@192.168.129.85;transport=tcp>
Content-Length: 0


[Jan 26 11:53:23] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP request (493 bytes) from UDP:10.10.54.45:5241 --->
REFER sip:asterisk@10.10.54.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.54.45:5241;branch=z9hG4bK-393434-5710f67168c6dc868c8eef9a206c0494
CSeq: 1 REFER
From: <sip:9833@10.10.54.45>;tag=Agent-2
To: <sip:+420112855219@10.113.157.108>;tag=02460ee6-0800-408d-a406-6b9ba32d7780
Call-ID: 198abaf5-9dc7-4515-919d-6eef3c64a01b
Max-Forwards: 70
Contact: <sip:9833@10.10.54.45:5241;transport=udp>;expires=479
Refer-To: <sip:+420115712700@10.10.54.20>
Refer-Sub: false
Content-Length: 0


[Jan 26 11:53:23] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (600 bytes) to UDP:10.10.54.45:5241 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.54.45:5241;rport=5241;received=10.10.54.45;branch=z9hG4bK-393434-5710f67168c6dc868c8eef9a206c0494
Call-ID: 198abaf5-9dc7-4515-919d-6eef3c64a01b
From: <sip:9833@10.10.54.45>;tag=Agent-2
To: <sip:+420112855219@10.113.157.108>;tag=02460ee6-0800-408d-a406-6b9ba32d7780
CSeq: 1 REFER
Refer-Sub: false
Contact: <sip:asterisk@10.10.54.20:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: asterisk1
Content-Length:  0


[Jan 26 11:53:23] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (1113 bytes) to TCP:192.168.129.85:18116 --->
INVITE sip:sgc_c@192.168.129.85;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj6a5ccf02-c219-4d79-a451-e84347f721a2;alias
From: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
To: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
Contact: <sip:10.10.54.20:5060;transport=TCP>
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 18136 INVITE
Route: <sip:192.168.129.85:18116;transport=TCP;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Require: 100rel
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: asterisk1
Content-Type: application/sdp
Content-Length:   245

[Jan 26 11:53:23] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (1328 bytes) to TCP:192.168.129.85:5060 --->
INVITE sip:+420115712700@domain1 SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj21e29e61-73c2-44e8-bbc0-648baf954342;alias
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
To: <sip:+420115712700@domain1>
Contact: <sip:+420119001486@10.10.54.20:5060;transport=TCP>
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24781 INVITE
Route: <sip:192.168.129.85;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:+420112809833@domain1>
History-Info: <sip:+420112809833@domain1;user=phone?Reason=Diversion%3btext%3d%22deflection%22>;answered;index=1,<sip:+420115712700@domain1;user=phone;cause=480>;index=1.1
P-Early-Media: Supported
Max-Forwards: 70
User-Agent: asterisk1
Content-Type: application/sdp
Content-Length:   256

[Jan 26 11:53:23] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (576 bytes) from TCP:192.168.129.85:18116 --->
SIP/2.0 407 Proxy Authentication Required 02035033D
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj6a5ccf02-c219-4d79-a451-e84347f721a2;alias
To: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
From: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 18136 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",realm="domain1",algorithm=MD5,qop="auth"


[Jan 26 11:53:23] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (547 bytes) to TCP:192.168.129.85:18116 --->
ACK sip:sgc_c@192.168.129.85;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj6a5ccf02-c219-4d79-a451-e84347f721a2;alias
From: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
To: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 18136 ACK
Route: <sip:192.168.129.85:18116;transport=TCP;lr>
Max-Forwards: 70
User-Agent: asterisk1
Content-Length:  0


[Jan 26 11:53:23] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (1443 bytes) to TCP:192.168.129.85:18116 --->
INVITE sip:sgc_c@192.168.129.85;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj6d259c1a-d27f-4e4e-b595-b8433557d750;alias
From: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
To: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
Contact: <sip:10.10.54.20:5060;transport=TCP>
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 18137 INVITE
Route: <sip:192.168.129.85:18116;transport=TCP;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Require: 100rel
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: asterisk1
Proxy-Authorization: Digest username="+420119001486@domain1", realm="domain1", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:sgc_c@192.168.129.85;transport=tcp", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5, cnonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   245

[Jan 26 11:53:23] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (550 bytes) from TCP:192.168.129.85:5060 --->
SIP/2.0 407 Proxy Authentication Required 02035033D
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj21e29e61-73c2-44e8-bbc0-648baf954342;alias
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_2781e9bf031d67d0fa52074275314a
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24781 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",realm="domain1",algorithm=MD5,qop="auth"


[Jan 26 11:53:23] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (506 bytes) to TCP:192.168.129.85:5060 --->
ACK sip:+420115712700@domain1 SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj21e29e61-73c2-44e8-bbc0-648baf954342;alias
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_2781e9bf031d67d0fa52074275314a
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24781 ACK
Route: <sip:192.168.129.85;lr>
Max-Forwards: 70
User-Agent: asterisk1
Content-Length:  0


[Jan 26 11:53:23] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (1682 bytes) to TCP:192.168.129.85:5060 --->
INVITE sip:+420115712700@domain1 SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj2efc966e-fc74-45cf-a066-f183e34a4e0d;alias
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
To: <sip:+420115712700@domain1>
Contact: <sip:+420119001486@10.10.54.20:5060;transport=TCP>
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24782 INVITE
Route: <sip:192.168.129.85:5060;transport=TCP;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:+420112809833@domain1>
History-Info: <sip:+420112809833@domain1;user=phone?Reason=Diversion%3btext%3d%22deflection%22>;answered;index=1,<sip:+420115712700@domain1;user=phone;cause=480>;index=1.1
P-Early-Media: Supported
Max-Forwards: 70
User-Agent: asterisk1
Proxy-Authorization: Digest username="+420119001486@domain1", realm="domain1", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:+420115712700@domain1", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5, cnonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   256

[Jan 26 11:53:24] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (1414 bytes) from TCP:192.168.129.85:18116 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj6d259c1a-d27f-4e4e-b595-b8433557d750;alias
To: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
From: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 18137 INVITE
Contact: <sip:sgc_c@192.168.129.85;transport=tcp>
Require: timer
Session-Expires: 1800;refresher=uas
Supported: timer
Content-Type: application/sdp
Content-Length: 162
Authentication-Info: qop=auth,rspauth="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",cnonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",nc=00000001
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY

[Jan 26 11:53:24] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (547 bytes) to TCP:192.168.129.85:18116 --->
ACK sip:sgc_c@192.168.129.85;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPjd58efc63-a5f3-4448-909e-d4d3e2622c43;alias
From: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
To: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 18137 ACK
Route: <sip:192.168.129.85:18116;transport=TCP;lr>
Max-Forwards: 70
User-Agent: asterisk1
Content-Length:  0


[Jan 26 11:53:24] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (346 bytes) from TCP:192.168.129.85:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj2efc966e-fc74-45cf-a066-f183e34a4e0d;alias
To: <sip:+420115712700@domain1>
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24782 INVITE
Content-Length: 0


[Jan 26 11:53:28] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (1281 bytes) from TCP:192.168.129.85:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj2efc966e-fc74-45cf-a066-f183e34a4e0d;alias
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24782 INVITE
Contact: <sip:sgc_c@192.168.129.85;transport=tcp>
Record-Route: <sip:192.168.129.85;transport=tcp;lr>
History-Info: <sip:+420112809833@domain1;user=phone?reason=Diversion%3btext%3d%22deflection%22>;answered;index=1
History-Info: <sip:+420115712700@domain1;user=phone;cause=480>;index=1.1
P-Asserted-Identity: "Pool User Deutsche Telekom" <sip:+420112809833@deutschland-lan.de;user=phone>
P-Early-Media: inactive, gated
Privacy: none
Require: 100rel
RSeq: 584003761
Supported: timer
Content-Type: application/sdp
Content-Length: 231
Session: Media
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE

[Jan 26 11:53:28] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (548 bytes) to TCP:192.168.129.85:5060 --->
PRACK sip:sgc_c@192.168.129.85;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPjf867904c-d9c1-4953-8785-527e4f16d52c;alias
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24783 PRACK
Route: <sip:192.168.129.85:5060;transport=TCP;lr>
RAck: 584003761 24782 INVITE
Max-Forwards: 70
User-Agent: asterisk1
Content-Length:  0


[Jan 26 11:53:28] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (411 bytes) from TCP:192.168.129.85:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPjf867904c-d9c1-4953-8785-527e4f16d52c;alias
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24783 PRACK
P-Early-Media: gated, inactive
Content-Length: 0


[Jan 26 11:53:28] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP request (864 bytes) from TCP:192.168.129.85:18116 --->
UPDATE sip:+420119001486@10.10.54.20:5060;transport=tcp SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/TCP 192.168.129.85:5060;branch=z9hG4bKg3Zqkv7ilknbzgj8bn7o5wpb5it7pqbgd
To: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
From: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 584001492 UPDATE
Contact: <sip:sgc_c@192.168.129.85;transport=tcp>
Min-Se: 900
P-Early-Media: inactive, gated
Privacy: none
Session-Expires: 1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 219

[Jan 26 11:53:28] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (982 bytes) to TCP:192.168.129.85:18116 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.129.85:5060;rport=18116;received=192.168.129.85;branch=z9hG4bKg3Zqkv7ilknbzgj8bn7o5wpb5it7pqbgd
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
From: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
To: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
CSeq: 584001492 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:+420119001486@10.10.54.20:5060;transport=TCP>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: asterisk1
Content-Type: application/sdp
Content-Length:   232


[Jan 26 11:53:28] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (985 bytes) from TCP:192.168.129.85:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj2efc966e-fc74-45cf-a066-f183e34a4e0d;alias
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24782 INVITE
Contact: <sip:sgc_c@192.168.129.85;transport=tcp>
Record-Route: <sip:192.168.129.85;transport=tcp;lr>
History-Info: <sip:+420112809833@domain1;user=phone?reason=Diversion%3btext%3d%22deflection%22>;answered;index=1
History-Info: <sip:+420115712700@domain1;user=phone;cause=480>;index=1.1
P-Asserted-Identity: "Pool User Deutsche Telekom" <sip:+420112809833@deutschland-lan.de;user=phone>
P-Early-Media: sendrecv
Privacy: none
Require: 100rel
RSeq: 584003762
Supported: timer
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE


[Jan 26 11:53:28] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (548 bytes) to TCP:192.168.129.85:5060 --->
PRACK sip:sgc_c@192.168.129.85;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj9c47f545-683d-408e-9c88-25e5537a960c;alias
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24784 PRACK
Route: <sip:192.168.129.85:5060;transport=TCP;lr>
RAck: 584003762 24782 INVITE
Max-Forwards: 70
User-Agent: asterisk1
Content-Length:  0


[Jan 26 11:53:29] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (404 bytes) from TCP:192.168.129.85:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj9c47f545-683d-408e-9c88-25e5537a960c;alias
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24784 PRACK
P-Early-Media: sendrecv
Content-Length: 0


[Jan 26 11:53:30] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP request (869 bytes) from TCP:192.168.129.85:18116 --->
UPDATE sip:+420119001486@10.10.54.20:5060;transport=tcp SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/TCP 192.168.129.85:5060;branch=z9hG4bKg3Zqkv7ixltdfqrqtwgiqjse8wpifgwcm
To: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
From: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 584001493 UPDATE
Contact: <sip:sgc_c@192.168.129.85;transport=tcp>
Min-Se: 900
P-Early-Media: sendrecv
Privacy: none
Session-Expires: 1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 231


[Jan 26 11:53:30] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (982 bytes) to TCP:192.168.129.85:18116 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.129.85:5060;rport=18116;received=192.168.129.85;branch=z9hG4bKg3Zqkv7ixltdfqrqtwgiqjse8wpifgwcm
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
From: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
To: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
CSeq: 584001493 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:+420119001486@10.10.54.20:5060;transport=TCP>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: asterisk1
Content-Type: application/sdp
Content-Length:   232


[Jan 26 11:53:30] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (1650 bytes) from TCP:192.168.129.85:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj2efc966e-fc74-45cf-a066-f183e34a4e0d;alias
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24782 INVITE
Contact: <sip:sgc_c@192.168.129.85;transport=tcp>
Record-Route: <sip:192.168.129.85;transport=tcp;lr>
History-Info: <sip:+420112809833@domain1;user=phone?reason=Diversion%3btext%3d%22deflection%22>;answered;index=1
History-Info: <sip:+420115712700@domain1;user=phone;cause=480>;index=1.1
P-Asserted-Identity: "Pool User Deutsche Telekom" <sip:+420112809833@deutschland-lan.de;user=phone>
Privacy: none
Require: timer
Session-Expires: 1800;refresher=uas
Supported: timer
Content-Length: 0
Session-ID: de5e9bade78c84ce7f9d43f184d3ccf2;remote=a5f4acb1bfdbd18c768a4231b0991a8b
Authentication-Info: qop=auth,rspauth="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",cnonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",nc=00000001
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE

[Jan 26 11:53:30] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (514 bytes) to TCP:192.168.129.85:5060 --->
ACK sip:sgc_c@192.168.129.85;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj100a6aa8-b838-44ce-8271-bbfc693f959e;alias
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24782 ACK
Route: <sip:192.168.129.85:5060;transport=TCP;lr>
Max-Forwards: 70
User-Agent: asterisk1
Content-Length:  0


[Jan 26 11:53:43] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP request (455 bytes) from TCP:192.168.129.85:18116 --->
BYE sip:10.10.54.20:5060;transport=tcp SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/TCP 192.168.129.85:5060;branch=z9hG4bKg3Zqkv7i1jvd3q7vl3lsspeuo11yqxkdv
To: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
Call-ID: BW1253213522601241916308192@10.2.137.167
CSeq: 583992423 BYE
Content-Length: 0


[Jan 26 11:53:43] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (470 bytes) to TCP:192.168.129.85:18116 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.129.85:5060;rport=18116;received=192.168.129.85;branch=z9hG4bKg3Zqkv7i1jvd3q7vl3lsspeuo11yqxkdv
Call-ID: BW1253213522601241916308192@10.2.137.167
From: <sip:+420112855219@domain1;user=phone>;tag=h7g4Esbg_467206822-1706270001352-
To: <sip:+420112809833@domain1;user=phone>;tag=cdcee971-d330-47ad-9ed2-09d85c926260;cscf
CSeq: 583992423 BYE
Server: asterisk1
Content-Length:  0


[Jan 26 11:53:43] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP request (514 bytes) to TCP:192.168.129.85:5060 --->
BYE sip:sgc_c@192.168.129.85;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj5c227c32-8a15-4c44-81d6-d69f6043ddde;alias
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24785 BYE
Route: <sip:192.168.129.85:5060;transport=TCP;lr>
Max-Forwards: 70
User-Agent: asterisk1
Content-Length:  0


[Jan 26 11:53:43] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP response (377 bytes) from TCP:192.168.129.85:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.54.20:5060;branch=z9hG4bKPj5c227c32-8a15-4c44-81d6-d69f6043ddde;alias
To: <sip:+420115712700@domain1>;tag=h7g4Esbg_1429980229-1706270024034
From: <sip:+420112855219@domain1>;tag=f158bf3f-4d67-4ede-9f2e-67f0976a9bf2
Call-ID: 6a51d202-ee47-4029-ad8e-08a762d06dd1
CSeq: 24785 BYE
Content-Length: 0


[Jan 26 11:53:59] VERBOSE[244913] res_pjsip_logger.c: <--- Received SIP request (365 bytes) from UDP:10.10.54.45:5241 --->
BYE sip:asterisk@10.10.54.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.54.45:5241;branch=z9hG4bK-393434-bdb3fd1725a3ba37b121f7d5d440061e
CSeq: 1 BYE
From: <sip:9833@10.10.54.45>;tag=Agent-2
To: <sip:+420112855219@10.113.157.108>;tag=02460ee6-0800-408d-a406-6b9ba32d7780
Call-ID: 198abaf5-9dc7-4515-919d-6eef3c64a01b
Max-Forwards: 70
Content-Length: 0


[Jan 26 11:53:59] VERBOSE[843077] res_pjsip_logger.c: <--- Transmitting SIP response (390 bytes) to UDP:10.10.54.45:5241 --->
SIP/2.0 500 Invalid CSeq
Via: SIP/2.0/UDP 10.10.54.45:5241;rport=5241;received=10.10.54.45;branch=z9hG4bK-393434-bdb3fd1725a3ba37b121f7d5d440061e
Call-ID: 198abaf5-9dc7-4515-919d-6eef3c64a01b
From: <sip:9833@10.10.54.45>;tag=Agent-2
To: <sip:+420112855219@10.113.157.108>;tag=02460ee6-0800-408d-a406-6b9ba32d7780
CSeq: 1 BYE
Server: asterisk1
Content-Length:  0
5

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.