Asterisk & PJSIP issue with TLS

mladenm · October 19, 2017, 10:02am

Hi

We think we need some help with our Asterisk server.

We are using Asterisk 13.9.1 with Pjproject 2.5.5 on Ubuntu 16.04.

Server is located in the cloud, and test clients are on the local WiFi, behind the same router. We are, mostly successfully, making TLS calls between two clients.

However, we have been experiencing some problems, and we think we have traced their root to the specific invite message. It might be a configuration issue, and we would appreciate any help with resolving it.

We believe that source of the problems is “From” line in INVITE message forwarded to the callee, that contains “sip” , instead of “sips”, address. In further course of the call, it likely propagates to “To” headers, and eventually ends up in “Contact” header, which causes call to end due “SIPS Required” error. In our use case this happens when a callee tries to put call on hold.

This is the INVITE message that is received by asterisk from the caller:

[Oct 18 15:47:13] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP request (1630 bytes) from TLS:217.169.223.250:50986 --->
INVITE sips:zz2867@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Max-Forwards: 70
From: sips:yy0206@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:zz2867@xxx.xxx.xxx.xxx
Contact: <sips:yy0206@217.169.223.250:50986;transport=TLS;ob>
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12748 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Authorization: *******************************************************
Content-Type: application/sdp
Content-Length:   650

v=0
o=- 3717323230 3717323230 IN IP4 192.168.15.207
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 96
c=IN IP4 192.168.15.207
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.207
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=crypto:1 *******************************************************
a=crypto:2 *******************************************************
a=crypto:3 *******************************************************
a=crypto:4 *******************************************************

This is the INVITE message forwarded to the callee (note From line):

[Oct 18 15:47:14] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP request (1048 bytes) to TLS:217.169.223.250:43122 --->
INVITE sips:zz2867@217.169.223.250:43122;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
From: <sip:yy0206@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:zz2867@217.169.223.250;ob>
Contact: <sips:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 31285 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REGISTER, REFER
Supported: timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: VoIPServerBeta
Content-Type: application/sdp
Content-Length:   339

v=0
o=- 54900247 54900247 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 0 101
a=crypto:1 *******************************************************
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

There are two defined transports:

[transport-udp]
type=transport
bind=0.0.0.0:5060
protocol=udp

and

[transport-tls]
type=transport
bind=0.0.0.0:5060
protocol=tls
cert_file=***
priv_key_file=***
password=***
cipher=***
method=tlsv1
local_net=172.31.1.0/255.255.255.0
external_media_address=xxx.xxx.xxx.xxx
external_signaling_address=xxx.xxx.xxx.xxx
external_signaling_port=5060

(We are aware that default port for TLS is 5061. 5060 also works, and binding to 5061 didn’t help. Neither did removing UDP transport.)

All users have the same configuration. This is the example:

Endpoint:  yy0206                                        Not in use    0 of inf
     InAuth:  yy0206/yy0206
        Aor:  yy0206                                      1
      Contact:  yy0206/sips:yy0206@217.169.223.250 3d00261b2b Unknown         nan
   Identify:  yy0206/yy0206
        Match: 127.0.0.1/32


 ParameterName                      : ParameterValue
 =========================================================
 100rel                             : no
 accountcode                        : 
 aggregate_mwi                      : true
 allow                              : (gsm|ulaw)
 allow_subscribe                    : true
 allow_transfer                     : true
 aors                               : yy0206
 auth                               : yy0206
 bind_rtp_to_media_address          : false
 call_group                         : 
 callerid                           : <unknown>
 callerid_privacy                   : allowed_not_screened
 callerid_tag                       : 
 connected_line_method              : invite
 context                            : voip_test
 cos_audio                          : 0
 cos_video                          : 0
 device_state_busy_at               : 0
 direct_media                       : false
 direct_media_glare_mitigation      : none
 direct_media_method                : invite
 disable_direct_media_on_nat        : false
 dtls_ca_file                       : 
 dtls_ca_path                       : 
 dtls_cert_file                     : 
 dtls_cipher                        : 
 dtls_fingerprint                   : XXX
 dtls_private_key                   : 
 dtls_rekey                         : 0
 dtls_setup                         : active
 dtls_verify                        : No
 dtmf_mode                          : rfc4733
 fax_detect                         : false
 force_avp                          : false
 force_rport                        : true
 from_domain                        : 
 from_user                          : 
 g726_non_standard                  : false
 ice_support                        : false
 identify_by                        : username
 inband_progress                    : false
 language                           : 
 mailboxes                          : 
 media_address                      : 
 media_encryption                   : sdes
 media_encryption_optimistic        : false
 media_use_received_transport       : false
 message_context                    : 
 moh_suggest                        : default
 mwi_from_user                      : 
 mwi_subscribe_replaces_unsolicited : false
 named_call_group                   : 
 named_pickup_group                 : 
 one_touch_recording                : false
 outbound_auth                      : 
 outbound_proxy                     : 
 pickup_group                       : 
 record_off_feature                 : automixmon
 record_on_feature                  : automixmon
 rewrite_contact                    : false
 rpid_immediate                     : false
 rtp_engine                         : asterisk
 rtp_ipv6                           : false
 rtp_keepalive                      : 0
 rtp_symmetric                      : true
 rtp_timeout                        : 0
 rtp_timeout_hold                   : 0
 sdp_owner                          : -
 sdp_session                        : Asterisk
 send_diversion                     : true
 send_pai                           : false
 send_rpid                          : false
 set_var                            : 
 srtp_tag_32                        : false
 sub_min_expiry                     : 0
 t38_udptl                          : false
 t38_udptl_ec                       : none
 t38_udptl_ipv6                     : false
 t38_udptl_maxdatagram              : 0
 t38_udptl_nat                      : false
 timers                             : yes
 timers_min_se                      : 90
 timers_sess_expires                : 1800
 tone_zone                          : 
 tos_audio                          : 0
 tos_video                          : 0
 transport                          : 
 trust_id_inbound                   : false
 trust_id_outbound                  : false
 use_avpf                           : false
 use_ptime                          : true
 user_eq_phone                      : false
 voicemail_extension                :

We would very much appreciate any help with this issue.

Thank you,
Mladen.

jcolp · October 19, 2017, 10:08am

Each leg is independent from the other, the caller has no impact on the callee signaling to that degree. You’ll need to provide a full SIP log showing the failure.

mladenm · October 19, 2017, 12:19pm

Hi jcolp,

Here is log from asterisk server. Clients are Android phones, using pjsip2 library v2.5.5. Call is connected and media is flowing without problems. Problem occurs the moment callee tries to put call on hold (15:47:22 in server log). Use case when caller tries to put call on hold works without problems.

This is sever log:

[Oct 18 15:47:10] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP request (1312 bytes) from TLS:217.169.223.250:50986 --->
INVITE sips:yy867@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d;alias
Max-Forwards: 70
From: sips:zz867@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:yy867@xxx.xxx.xxx.xxx
Contact: <sips:zz867@217.169.223.250:50986;transport=TLS;ob>
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12747 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   650

v=0
o=- 3717323230 3717323230 IN IP4 192.168.15.207
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 96
c=IN IP4 192.168.15.207
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.207
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=crypto: ***********************
a=crypto: ***********************
a=crypto: ***********************
a=crypto: ***********************

[Oct 18 15:47:11] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP response (596 bytes) to TLS:217.169.223.250:50986 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 217.169.223.250:50986;rport=50986;received=217.169.223.250;branch=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: <sips:yy867@xxx.xxx.xxx.xxx>;tag=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d
CSeq: 12747 INVITE
WWW-Authenticate: ***********************
Server: ServerBeta
Content-Length:  0


[Oct 18 15:47:13] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP request (421 bytes) from TLS:217.169.223.250:50986 --->
ACK sips:yy867@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d;alias
Max-Forwards: 70
From: sips:zz867@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:yy867@xxx.xxx.xxx.xxx;tag=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12747 ACK
Content-Length:  0


[Oct 18 15:47:13] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP request (1630 bytes) from TLS:217.169.223.250:50986 --->
INVITE sips:yy867@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Max-Forwards: 70
From: sips:zz867@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:yy867@xxx.xxx.xxx.xxx
Contact: <sips:zz867@217.169.223.250:50986;transport=TLS;ob>
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12748 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Authorization: ***********************
Content-Type: application/sdp
Content-Length:   650

v=0
o=- 3717323230 3717323230 IN IP4 192.168.15.207
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 96
c=IN IP4 192.168.15.207
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.207
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=crypto: ***********************
a=crypto: ***********************
a=crypto: ***********************
a=crypto: ***********************

[Oct 18 15:47:14] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP response (393 bytes) to TLS:217.169.223.250:50986 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 217.169.223.250:50986;rport=50986;received=217.169.223.250;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: <sips:yy867@xxx.xxx.xxx.xxx>
CSeq: 12748 INVITE
Server: ServerBeta
Content-Length:  0


[Oct 18 15:47:14] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP request (1048 bytes) to TLS:217.169.223.250:43122 --->
INVITE sips:yy867@217.169.223.250:43122;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>
Contact: <sips:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 31285 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REGISTER, REFER
Supported: timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: ServerBeta
Content-Type: application/sdp
Content-Length:   339

v=0
o=- 54900247 54900247 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 0 101
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[Oct 18 15:47:14] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP response (360 bytes) from TLS:217.169.223.250:43122 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>
CSeq: 31285 INVITE
Content-Length:  0


[Oct 18 15:47:14] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP response (569 bytes) from TLS:217.169.223.250:43122 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
CSeq: 31285 INVITE
Contact: <sips:yy867@217.169.223.250:43122;transport=TLS;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


[Oct 18 15:47:14] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP response (587 bytes) to TLS:217.169.223.250:50986 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 217.169.223.250:50986;rport=50986;received=217.169.223.250;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: <sips:yy867@xxx.xxx.xxx.xxx>;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
CSeq: 12748 INVITE
Server: ServerBeta
Contact: <sips:xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REGISTER, REFER
Content-Length:  0


[Oct 18 15:47:16] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP response (1052 bytes) from TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
CSeq: 31285 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sips:yy867@217.169.223.250:43122;transport=TLS;ob>
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   353

v=0
o=- 3717323234 3717323235 IN IP4 192.168.15.218
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 101
c=IN IP4 192.168.15.218
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.218
a=sendrecv
a=rtpmap:3 GSM/8000
a=crypto: ***********************
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Oct 18 15:47:16] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP request (474 bytes) to TLS:217.169.223.250:43122 --->
ACK sips:yy867@217.169.223.250:43122;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPj0e57ffbc-a8e9-402b-905f-4786c48ac3de;alias
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 31285 ACK
Max-Forwards: 70
User-Agent: ServerBeta
Content-Length:  0


[Oct 18 15:47:16] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP response (1025 bytes) to TLS:217.169.223.250:50986 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:50986;rport=50986;received=217.169.223.250;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: <sips:yy867@xxx.xxx.xxx.xxx>;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
CSeq: 12748 INVITE
Server: ServerBeta
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REGISTER, REFER
Contact: <sips:xxx.xxx.xxx.xxx:5060;transport=TLS>
Supported: timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   316

v=0
o=- 3717323230 3717323232 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6026 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:300
a=sendrecv

[Oct 18 15:47:17] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP request (413 bytes) from TLS:217.169.223.250:50986 --->
ACK sips:xxx.xxx.xxx.xxx:5060;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPjadfe3536-0f6c-4a70-8ed4-dc665d4ceaf0;alias
Max-Forwards: 70
From: sips:zz867@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:yy867@xxx.xxx.xxx.xxx;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12748 ACK
Content-Length:  0

[Oct 18 15:47:22] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP request (1079 bytes) from TLS:217.169.223.250:43122 --->
INVITE sips:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:43122;rport;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Max-Forwards: 70
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
Contact: <sips:yy867@217.169.223.250:43122;transport=TLS;ob>
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 5159 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length:   343

v=0
o=- 3717323234 3717323236 IN IP4 192.168.15.218
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 96
c=IN IP4 0.0.0.0
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.218
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=crypto: ***********************
a=inactive

[Oct 18 15:47:22] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:22] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP request (471 bytes) from TLS:217.169.223.250:43122 --->
BYE sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:43122;rport;branch=z9hG4bKPja369cb16-e0a3-496d-b9b3-165fb63dd4dc;alias
Max-Forwards: 70
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 5160 BYE
Warning: 381 localhost "SIPS Required"
Content-Length:  0


[Oct 18 15:47:22] VERBOSE[21718] res_pjsip_logger.c: <--- Transmitting SIP response (433 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPja369cb16-e0a3-496d-b9b3-165fb63dd4dc;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5160 BYE
Server: ServerBeta
Content-Length:  0


[Oct 18 15:47:22] VERBOSE[22168] res_pjsip_logger.c: <--- Transmitting SIP request (491 bytes) to TLS:217.169.223.250:50986 --->
BYE sips:zz867@217.169.223.250:50986;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPjd124dfc8-2048-46d0-8741-9878869a129f;alias
From: <sips:yy867@xxx.xxx.xxx.xxx>;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
To: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 1217 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ServerBeta
Content-Length:  0


[Oct 18 15:47:22] VERBOSE[21198] res_pjsip_logger.c: <--- Received SIP response (386 bytes) from TLS:217.169.223.250:50986 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bKPjd124dfc8-2048-46d0-8741-9878869a129f;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:yy867@xxx.xxx.xxx.xxx>;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
To: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
CSeq: 1217 BYE
Content-Length:  0


[Oct 18 15:47:22] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:23] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:25] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:29] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:33] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:37] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:41] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


[Oct 18 15:47:45] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:49] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

[Oct 18 15:47:53] VERBOSE[21198] res_pjsip_logger.c: <--- Transmitting SIP response (1034 bytes) to TLS:217.169.223.250:43122 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

Thank you for your help.

mladenm · October 19, 2017, 12:20pm

Here are callee and caller logs.

Callee:

INVITE sips:yy867@217.169.223.250:43122;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>
Contact: <sips:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 31285 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REGISTER, REFER
Supported: timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: ServerBeta
Content-Type: application/sdp
Content-Length:   339

v=0
o=- 54900247 54900247 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 0 101
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


SIP/2.0 100 Trying
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>
CSeq: 31285 INVITE
Content-Length:  0


SIP/2.0 180 Ringing
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
CSeq: 31285 INVITE
Contact: <sips:yy867@217.169.223.250:43122;transport=TLS;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


SIP/2.0 200 OK
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bKPj76d12694-8316-42c6-9c16-737dc1ce9c5d;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
CSeq: 31285 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sips:yy867@217.169.223.250:43122;transport=TLS;ob>
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   353

v=0
o=- 3717323234 3717323235 IN IP4 192.168.15.218
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 101
c=IN IP4 192.168.15.218
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.218
a=sendrecv
a=rtpmap:3 GSM/8000
a=crypto: ***********************
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


ACK sips:yy867@217.169.223.250:43122;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPj0e57ffbc-a8e9-402b-905f-4786c48ac3de;alias
From: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
To: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 31285 ACK
Max-Forwards: 70
User-Agent: ServerBeta
Content-Length:  0


INVITE sips:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:43122;rport;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Max-Forwards: 70
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
Contact: <sips:yy867@217.169.223.250:43122;transport=TLS;ob>
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 5159 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length:   343

v=0
o=- 3717323234 3717323236 IN IP4 192.168.15.218
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 96
c=IN IP4 0.0.0.0
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.218
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=crypto: ***********************
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


BYE sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:43122;rport;branch=z9hG4bKPja369cb16-e0a3-496d-b9b3-165fb63dd4dc;alias
Max-Forwards: 70
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
CSeq: 5160 BYE
Warning: 381 localhost "SIPS Required"
Content-Length:  0


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPja369cb16-e0a3-496d-b9b3-165fb63dd4dc;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5160 BYE
Server: ServerBeta
Content-Length:  0


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:43122;rport=43122;received=217.169.223.250;branch=z9hG4bKPj9f57318c-5319-4f07-a805-f9c4967d31ee;alias
Call-ID: 6c9e38cb-3515-421a-ae00-40636f1912ac
From: <sips:yy867@217.169.223.250;ob>;tag=4228d6b1-083a-4c0b-bbf8-510987e62649
To: <sip:zz867@172.31.1.100>;tag=a2953819-10ab-4673-ad5f-ba31bf3d9d60
CSeq: 5159 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Server: ServerBeta
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 54900247 54900248 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6016 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=inactive

Caller:

INVITE sips:yy867@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d;alias
Max-Forwards: 70
From: sips:zz867@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:yy867@xxx.xxx.xxx.xxx
Contact: <sips:zz867@217.169.223.250:50986;transport=TLS;ob>
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12747 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   650

v=0
o=- 3717323230 3717323230 IN IP4 192.168.15.207
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 96
c=IN IP4 192.168.15.207
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.207
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=crypto: ***********************
a=crypto: ***********************
a=crypto: ***********************
a=crypto: ***********************


SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 217.169.223.250:50986;rport=50986;received=217.169.223.250;branch=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: <sips:yy867@xxx.xxx.xxx.xxx>;tag=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d
CSeq: 12747 INVITE
WWW-Authenticate: ***********************
Server: ServerBeta
Content-Length:  0


ACK sips:yy867@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d;alias
Max-Forwards: 70
From: sips:zz867@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:yy867@xxx.xxx.xxx.xxx;tag=z9hG4bKPj230fd747-f9b8-4c46-8d54-3b8aa2ae089d
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12747 ACK
Content-Length:  0


INVITE sips:yy867@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Max-Forwards: 70
From: sips:zz867@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:yy867@xxx.xxx.xxx.xxx
Contact: <sips:zz867@217.169.223.250:50986;transport=TLS;ob>
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12748 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Authorization: ***********************
Content-Type: application/sdp
Content-Length:   650

v=0
o=- 3717323230 3717323230 IN IP4 192.168.15.207
s=pjmedia
b=AS:30
t=0 0
m=audio 4000 RTP/SAVP 3 96
c=IN IP4 192.168.15.207
b=TIAS:13200
a=rtcp:4001 IN IP4 192.168.15.207
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=crypto: ***********************
a=crypto: ***********************
a=crypto: ***********************
a=crypto: ***********************


SIP/2.0 100 Trying
Via: SIP/2.0/TLS 217.169.223.250:50986;rport=50986;received=217.169.223.250;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: <sips:yy867@xxx.xxx.xxx.xxx>
CSeq: 12748 INVITE
Server: ServerBeta
Content-Length:  0


SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 217.169.223.250:50986;rport=50986;received=217.169.223.250;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: <sips:yy867@xxx.xxx.xxx.xxx>;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
CSeq: 12748 INVITE
Server: ServerBeta
Contact: <sips:xxx.xxx.xxx.xxx:5060;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REGISTER, REFER
Content-Length:  0


SIP/2.0 200 OK
Via: SIP/2.0/TLS 217.169.223.250:50986;rport=50986;received=217.169.223.250;branch=z9hG4bKPjfc4612c5-1684-465d-bc4e-3efe626e3f31;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: <sips:yy867@xxx.xxx.xxx.xxx>;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
CSeq: 12748 INVITE
Server: ServerBeta
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REGISTER, REFER
Contact: <sips:xxx.xxx.xxx.xxx:5060;transport=TLS>
Supported: timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   316

v=0
o=- 3717323230 3717323232 IN IP4 172.31.1.100
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 6026 RTP/SAVP 3 96
a=crypto: ***********************
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:300
a=sendrecv


ACK sips:xxx.xxx.xxx.xxx:5060;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 217.169.223.250:50986;rport;branch=z9hG4bKPjadfe3536-0f6c-4a70-8ed4-dc665d4ceaf0;alias
Max-Forwards: 70
From: sips:zz867@xxx.xxx.xxx.xxx;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
To: sips:yy867@xxx.xxx.xxx.xxx;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 12748 ACK
Content-Length:  0


BYE sips:zz867@217.169.223.250:50986;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPjd124dfc8-2048-46d0-8741-9878869a129f;alias
From: <sips:yy867@xxx.xxx.xxx.xxx>;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
To: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
CSeq: 1217 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ServerBeta
Content-Length:  0


SIP/2.0 200 OK
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bKPjd124dfc8-2048-46d0-8741-9878869a129f;alias
Call-ID: a62d94a3-17ba-4fd5-acb7-c3e6e9514f5a
From: <sips:yy867@xxx.xxx.xxx.xxx>;tag=be210663-0f2b-4f50-997f-4c5fcb9fc750
To: <sips:zz867@xxx.xxx.xxx.xxx>;tag=382ff707-70d7-43e5-ad8c-ca3319e3d124
CSeq: 1217 BYE
Content-Length:  0

jcolp · October 19, 2017, 12:23pm

So yes, the Contact is incorrect there. I’d suggest trying the latest version of Asterisk first (yours is about 1.5 years old) to ensure that it hasn’t been fixed already. After that if it’s not resolved please file an issue[1] with the above logs and configuration.

[1] https://issues.asterisk.org/jira

mladenm · October 19, 2017, 12:30pm

Hi,

We tried Asterisk versions above 13.9.1 but we had issue with the TLS on all of them. Is something changed in later versions regarding SSL that we have missed?

jcolp · October 19, 2017, 12:32pm

It’s possible. I don’t know off the top of my head, but before we accept an issue we require that the latest version be used.

mladenm · October 19, 2017, 12:36pm

Sure thing. Thanks for the help. I will see when our team can try to migrate to the latest Asterisk version.

traud · October 30, 2017, 9:34am

For a short-term workaround: Who is stating this ‘SIPS Required’ – Asterisk or your client? If it is Asterisk, go for the file /res/res_pjsip/config_system.c and add
pjsip_cfg()->endpt.disable_secure_dlg_check = PJ_TRUE;
into the function system_apply. After that, you have to re-make Asterisk.

Did that help? If yes, please, provide more (abstract) information about your call scenario, who is calling whom (product, version, manufacturer) and who is placing whom on hold.

Topic		Replies	Views
PJSIP.conf Try to call out Asterisk SIP	9	977	November 29, 2021
PJSIP NAT Issues Asterisk SIP	7	1094	January 31, 2020
Asterisk 13.5 - PJSIP problems with TLS Asterisk Support	8	2673	October 28, 2015
[Call-Forwarding] What's the problem? Asterisk SIP	2	617	February 19, 2019
Asterisk 13.3.2 - PJSIP strange behavior under TLS Asterisk Support	6	474	May 22, 2015

Asterisk & PJSIP issue with TLS

Related topics