Incoming call using TLS: 488 Not Acceptable Here

Hi
I’m trying to setup Asterisk and I have a problem for receiving a call.
I’m trying to call the phone number provided by my SIP provider from a mobile phone and, as mentioned in the title, I’m receiving: 488 Not Acceptable Here.
For the communication, I’m using TLS (cannot use TCP or UPD on port 5060).
I’m thinking a codec or encryption issue but I don’t know how to determine it.

Thanks for your support because, as I’m little bit lost with these multiple parameters…

Trace from Asterisk

<--- Received SIP request (1832 bytes) from TLS:2XX.XX.XX.XX:5061 --->
INVITE sip:+XXYYXXYYXX210@7Y.YY.YY.YY:5061;transport=TLS;line=gmycppc SIP/2.0
Via: SIP/2.0/TLS 2XX.XX.XX.XX:5061;branch=z9hG4bK3f31e8d4739be6d9ef11b7d8a4d5cb70.f0d2e0db
Record-Route: <sip:rtrunk.server.com;transport=tls;lr>
Max-Forwards: 50
To: <sip:+XXYYXXYYXX210@sip.com;user=phone>
From: <sip:+XXYYXXYYXXYY5@trunk.server.com;user=phone>;tag=b9bd8c47
Call-ID: 559a361b5ba8dcd9@8Z.ZZ.ZZ.ZZ
Contact: <sip:DYG+sozv+dQYyKt1B49ETNPoMg6seNRKnjFysrywk/WFjuWyeNbqLlGtLFk+daEZDUse@th1>
Supported: 100rel,histinfo,norefersub,uui
CSeq: 27335786 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, UPDATE
P-Asserted-Identity: <sip:+XXYYXXYYXXYY5@trunk.server.com;user=phone>
P-Called-Party-ID: <sip:+XXYYXXYYXX210@trunk.server.com;user=phone>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 858

v=0
o=- 0 0 IN IP4 2XX.XX.XX.XX
s=on transit
c=IN IP4 217.0.14.241
t=0 0
m=audio 17396 RTP/SAVP 8 96 97 98 99 100
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-set=0,1,2,3,4,5,6,7; mode-change-period=2; mode-change-neighbor=1; max-red=0
a=rtpmap:97 AMR/8000
a=fmtp:97 mode-set=0,2,4,7; mode-change-period=2; mode-change-neighbor=1; max-red=0
a=rtpmap:98 AMR/8000
a=fmtp:98 mode-set=0,1,2,3,4,5,6,7; max-red=0
a=rtpmap:99 AMR/8000
a=fmtp:99 mode-set=0,2,4; mode-change-period=2; mode-change-neighbor=1; max-red=0
a=rtpmap:100 telephone-event/8000
a=ptime:20
a=maxptime:30
a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:38eLn0d5r0LbezBcEEXY/zQtNRlUk3akC899use9
a=crypto:6 AES_CM_128_HMAC_SHA1_32 inline:KiWfRwoDTDip3nxMNDnGs/9Cs/4XSGvcLT19Z+n/
a=crypto:7 F8_128_HMAC_SHA1_80 inline:uyiIscXy+H6QuuTknvsfOYlh3Spq0bGYZqhrkzY9

  == Setting global variable 'SIPDOMAIN' to '7Y.YY.YY.YY'
<--- Transmitting SIP response (447 bytes) to TLS:2XX.XX.XX.XX:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 2XX.XX.XX.XX:5061;rport=5061;received=2XX.XX.XX.XX;branch=z9hG4bK3f31e8d4739be6d9ef11b7d8a4d5cb70.f0d2e0db
Record-Route: <sip:rtrunk.server.com;transport=tls;lr>
Call-ID: 559a361b5ba8dcd9@8Z.ZZ.ZZ.ZZ
From: <sip:+XXYYXXYYXXYY5@trunk.server.com;user=phone>;tag=b9bd8c47
To: <sip:+XXYYXXYYXX210@sip.com;user=phone>
CSeq: 27335786 INVITE
Server: Asterisk PBX 17.3.0
Content-Length:  0

<--- Transmitting SIP response (497 bytes) to TLS:2XX.XX.XX.XX:5061 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 2XX.XX.XX.XX:5061;rport=5061;received=2XX.XX.XX.XX;branch=z9hG4bK3f31e8d4739be6d9ef11b7d8a4d5cb70.f0d2e0db
Record-Route: <sip:rtrunk.server.com;transport=tls;lr>
Call-ID: 559a361b5ba8dcd9@8Z.ZZ.ZZ.ZZ
From: <sip:+XXYYXXYYXXYY5@trunk.server.com;user=phone>;tag=b9bd8c47
To: <sip:+XXYYXXYYXX210@sip.com;user=phone>;tag=BEFPG9v52EjfS5nuLoRNYBS5x.iReQrZ
CSeq: 27335786 INVITE
Server: Asterisk PBX 17.3.0
Content-Length:  0

<--- Received SIP request (431 bytes) from TLS:2XX.XX.XX.XX:5061 --->
ACK sip:+XXYYXXYYXX210@7Y.YY.YY.YY:5061;transport=TLS;line=gmycppc SIP/2.0
Via: SIP/2.0/TLS 2XX.XX.XX.XX:5061;branch=z9hG4bK3f31e8d4739be6d9ef11b7d8a4d5cb70.f0d2e0db
Max-Forwards: 50
To: <sip:+XXYYXXYYXX210@sip.com;user=phone>;tag=BEFPG9v52EjfS5nuLoRNYBS5x.iReQrZ
From: <sip:+XXYYXXYYXXYY5@trunk.server.com;user=phone>;tag=b9bd8c47
Call-ID: 559a361b5ba8dcd9@8Z.ZZ.ZZ.ZZ
CSeq: 27335786 ACK
Content-Length: 0

———————————————————————————
sip.conf (extract for incoming calls)

[ mytrunk-in]
type=endpoint
transport=transport-tls
context=incoming-calls
allow=!all,g722,alaw,ulaw
direct_media=yes
dtmf_mode=rfc4733
outbound_auth=trunk-auth
tos_audio=184
 
[mytrunk-in]
type=identify
endpoint=mytrunk-in
match=2XX.0.0.0/13

———————————————————————————
sip.conf

[general]
context=public                  ; Default context for incoming calls. Defaults to 'default'
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp                   ; Set the default transports.  The order determines the primary default transport.
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
tlsdontverifyserver=yes;
tlscapath=/etc/asterisk/keys/trustedcas
localnet=192.168.0.190/255.255.255.254
externaddr=7Y.YY.YY.YY:5061
nat=force_rport,comedia

———————————————————————————
extension.conf

[incoming-calls]
exten => _+XXXXXXXXXXXX0,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()

Hi,
Since I’ve posted my issue, I checked again the forums and, for most of the people,
“488” is supposed to means that there is no media compatibility.
Following the trace, it is requesting PCMA/8000. In my pjsip.conf, I’m allowing alaw and ulaw.
Did I miss something? Is it requesting another codec.

I’ve read also that using TLS may also bring others issues with Asterisk. As mentioned, unfortunately, I have to go with TLS. Can someone confirm is it working with Asterisk 17

Last point, I made a mistake in my first message. I’m using pjsip; so, the extract is from pjsip.conf

[ mytrunk-in]
type=endpoint
transport=transport-tls
context=incoming-calls
allow=!all,g722,alaw,ulaw
direct_media=yes
dtmf_mode=rfc4733
outbound_auth=trunk-auth
tos_audio=184

[mytrunk-in]
type=identify
endpoint=mytrunk-in
match=2XX.0.0.0/13

Any ideas how to find the origin of this error?

It is requesting SDES media encryption as well. Your endpoint is not configured for that, so the SDP negotiation fails since you are only supporting unencrypted. It is enabled by setting the “media_encryption” option to “sdes”.

As proposed, I’ve added this option but, unfortunately, I’m still getting the same error message…

[mytrunk-in]
type=endpoint
transport=transport-tls
context=incoming-calls
allow=!all,g722,alaw,ulaw
direct_media=yes
dtmf_mode=rfc4733
outbound_auth=trunk-auth
tos_audio=184
media_encryption=sdes

[mytrunk-in]
type=identify
endpoint=mytrunk-in
match=217.0.0.0/13


<--- Received SIP request (1830 bytes) from TLS:2XX.XX.XX.XX:5061 --->

INVITE sip:+XXYYXXYYXX210@7Y.YY.YY.YY:5061;transport=TLS;line=pksller SIP/2.0

Via: SIP/2.0/TLS 2XX.XX.XX.XX:5061;branch=z9hG4bK71612886c93a4481f19df8a2c6af7154.4656a31a

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Max-Forwards: 50

To: <sip:+XXYYXXYYXX210@server.com;user=phone>

From: <sip:+XXYYXXYYXXYY5@sip-trunk.server.com;user=phone>;tag=3a94d62a

Call-ID: cd42bab8f42ad83e@6Z.ZZ.ZZ.71

Contact: <sip:JQu5bCu8cwls2iFFl/M9JehjleIomS7rUEq4WBxkuQfT/7by1DfOAV95BlWc59IUy/5x@th1>

Supported: 100rel,histinfo,norefersub,uui

CSeq: 4389793 INVITE

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, UPDATE

P-Asserted-Identity: <sip:+XXYYXXYYXXYY5@sip-trunk.server.com;user=phone>

P-Called-Party-ID: <sip:+XXYYXXYYXX210@sip-trunk.server.com;user=phone>

Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 858

v=0

o=- 0 0 IN IP4 2XX.XX.XX.XX

s=on transit

c=IN IP4 217.0.14.130

t=0 0

m=audio 10710 RTP/SAVP 8 96 97 98 99 100

b=AS:80

a=rtpmap:8 PCMA/8000

a=rtpmap:96 AMR/8000

a=fmtp:96 mode-set=0,1,2,3,4,5,6,7; mode-change-period=2; mode-change-neighbor=1; max-red=0

a=rtpmap:97 AMR/8000

a=fmtp:97 mode-set=0,2,4,7; mode-change-period=2; mode-change-neighbor=1; max-red=0

a=rtpmap:98 AMR/8000

a=fmtp:98 mode-set=0,1,2,3,4,5,6,7; max-red=0

a=rtpmap:99 AMR/8000

a=fmtp:99 mode-set=0,2,4; mode-change-period=2; mode-change-neighbor=1; max-red=0

a=rtpmap:100 telephone-event/8000

a=ptime:20

a=maxptime:30

a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:TfW1a2lCtERA0ZHLoDNEL0h4dMmivgJn12qYuimP

a=crypto:6 AES_CM_128_HMAC_SHA1_32 inline:ff7Q/mF+oypfb9jLzXvu1LGalo/Nf+X9x3M2f1bD

a=crypto:7 F8_128_HMAC_SHA1_80 inline:9u92/Zx+Z6XIFHTSUSXLqLNvguKMIUNET/pO38qH

== Setting global variable 'SIPDOMAIN' to '7Y.YY.YY.YY'

<--- Transmitting SIP response (445 bytes) to TLS:2XX.XX.XX.XX:5061 --->

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 2XX.XX.XX.XX:5061;rport=5061;received=2XX.XX.XX.XX;branch=z9hG4bK71612886c93a4481f19df8a2c6af7154.4656a31a

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Call-ID: cd42bab8f42ad83e@6Z.ZZ.ZZ.71

From: <sip:+XXYYXXYYXXYY5@sip-trunk.server.com;user=phone>;tag=3a94d62a

To: <sip:+XXYYXXYYXX210@server.com;user=phone>

CSeq: 4389793 INVITE

Server: Asterisk PBX 17.3.0

Content-Length: 0

== Using SIP RTP Audio TOS bits 184

<--- Transmitting SIP response (495 bytes) to TLS:2XX.XX.XX.XX:5061 --->

SIP/2.0 488 Not Acceptable Here

Via: SIP/2.0/TLS 2XX.XX.XX.XX:5061;rport=5061;received=2XX.XX.XX.XX;branch=z9hG4bK71612886c93a4481f19df8a2c6af7154.4656a31a

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Call-ID: cd42bab8f42ad83e@6Z.ZZ.ZZ.71

From: <sip:+XXYYXXYYXXYY5@sip-trunk.server.com;user=phone>;tag=3a94d62a

To: <sip:+XXYYXXYYXX210@server.com;user=phone>;tag=3tveiSuV7FhHc-qBqjVS6-i7o.6A9BMU

CSeq: 4389793 INVITE

Server: Asterisk PBX 17.3.0

Content-Length: 0

<--- Received SIP request (429 bytes) from TLS:2XX.XX.XX.XX:5061 --->

ACK sip:+XXYYXXYYXX210@7Y.YY.YY.YY:5061;transport=TLS;line=pksller SIP/2.0

Via: SIP/2.0/TLS 2XX.XX.XX.XX:5061;branch=z9hG4bK71612886c93a4481f19df8a2c6af7154.4656a31a

Max-Forwards: 50

To: <sip:+XXYYXXYYXX210@server.com;user=phone>;tag=3tveiSuV7FhHc-qBqjVS6-i7o.6A9BMU

From: <sip:+XXYYXXYYXXYY5@sip-trunk.server.com;user=phone>;tag=3a94d62a

Call-ID: cd42bab8f42ad83e@6Z.ZZ.ZZ.71

CSeq: 4389793 ACK

Content-Length: 0

I assume you reloaded as well? Does “pjsip show endpoint mytrunk-in” confirm the option is set? Is the res_srtp module loaded?

yes - I’ve reloaded. Hereafter, the result of the command (media_encryption is set to sdes.
Now, regarding the module, it is not available.

module load res_srtp.so
Unable to load module res_srtp.so
Command ‘module load res_srtp.so’ failed.
[May 18 11:05:27] ERROR[3204]: loader.c:281 module_load_error: Error loading module ‘res_srtp.so’: /usr/lib/asterisk/modules/res_srtp.so: cannot open shared object file: No such file or directory

Is it not part of Asterisk? Compiled by default?
FYI, I rebuilt asterisk and got the source code from git (asterisk-17.3.0)…


 ParameterName                      : ParameterValue
 =========================================================
 100rel                             : yes
 accept_multiple_sdp_answers        : false
 accountcode                        :
 acl                                :
 aggregate_mwi                      : true
 allow                              : (g722|alaw|ulaw)
 allow_overlap                      : true
 allow_subscribe                    : true
 allow_transfer                     : true
 aors                               :
 asymmetric_rtp_codec               : false
 auth                               :
 bind_rtp_to_media_address          : false
 bundle                             : false
 call_group                         :
 callerid                           : <unknown>
 callerid_privacy                   : allowed_not_screened
 callerid_tag                       :
 connected_line_method              : invite
 contact_acl                        :
 context                            : incoming-calls
 cos_audio                          : 0
 cos_video                          : 0
 device_state_busy_at               : 0
 direct_media                       : true
 direct_media_glare_mitigation      : none
 direct_media_method                : invite
 disable_direct_media_on_nat        : false
 dtls_auto_generate_cert            : No
 dtls_ca_file                       :
 dtls_ca_path                       :
 dtls_cert_file                     :
 dtls_cipher                        :
 dtls_fingerprint                   : SHA-256
 dtls_private_key                   :
 dtls_rekey                         : 0
 dtls_setup                         : active
 dtls_verify                        : No
 dtmf_mode                          : rfc4733
 fax_detect                         : false
 fax_detect_timeout                 : 0
 follow_early_media_fork            : true
 force_avp                          : false
 force_rport                        : true
 from_domain                        :
 from_user                          :
 g726_non_standard                  : false
 ice_support                        : false
 identify_by                        : username,ip
 ignore_183_without_sdp             : false
 inband_progress                    : false
 incoming_mwi_mailbox               :
 language                           :
 mailboxes                          :
 max_audio_streams                  : 1
 max_video_streams                  : 1
 media_address                      :
 media_encryption                   : sdes
 media_encryption_optimistic        : false
 media_use_received_transport       : false
 message_context                    :
 moh_passthrough                    : false
 moh_suggest                        : default
 mwi_from_user                      :
 mwi_subscribe_replaces_unsolicited : no
 named_call_group                   :
 named_pickup_group                 :
 notify_early_inuse_ringing         : false
 one_touch_recording                : false
 outbound_auth                      : trunk-auth
 outbound_proxy                     :
 pickup_group                       :
 preferred_codec_only               : false
 record_off_feature                 : automixmon
 record_on_feature                  : automixmon
 refer_blind_progress               : true
 rewrite_contact                    : false
 rpid_immediate                     : false
 rtcp_mux                           : false
 rtp_engine                         : asterisk
 rtp_ipv6                           : false
 rtp_keepalive                      : 0
 rtp_symmetric                      : false
 rtp_timeout                        : 0
 rtp_timeout_hold                   : 0
 sdp_owner                          : -
 sdp_session                        : Asterisk
 send_connected_line                : yes
 send_diversion                     : true
 send_pai                           : false
 send_rpid                          : false
 set_var                            :
 srtp_tag_32                        : false
 sub_min_expiry                     : 0
 subscribe_context                  :
 suppress_q850_reason_headers       : false
 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                          : 184
 tos_video                          : 0
 transport                          : transport-tls
 trust_connected_line               : yes
 trust_id_inbound                   : false
 trust_id_outbound                  : false
 use_avpf                           : false
 use_ptime                          : false
 user_eq_phone                      : false
 voicemail_extension                :
 webrtc                             : no

The res_srtp module is part of Asterisk, however it requires the libsrtp library and development headers to be available in order to be built.

Thanks for your details. I’ve now rebuilt Asterisk with the required module and I’m going further.
I really appreciate your support…

When I’m calling my SIP number, the extension is ringing and I can accept the call.
Unfortunately, call is dropped and trace mentions new errors:
403 Forbidden (R403_REQUEST_NOT_ALLOWED)
followed by
481 Call/Transaction Does Not Exist


<--- Received SIP request (2152 bytes) from TLS:217.XX.XX.229:5061 --->

INVITE sip:+XXXXXXXXXX210@7X.XX.XX.156:5061;transport=TLS;line=ikilycm SIP/2.0

Via: SIP/2.0/TLS 217.XX.XX.229:5061;branch=z9hG4bK033a350245697d3777b63fa6cdd876fd.3acdd739

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Max-Forwards: 50

To: <sip:+XXXXXXXXXX210@server.com;user=phone>

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

Contact: <sip:PUOVKRq3x3KGmbVkZYHoGYG4b7dcwIDh7cLE/horRQvFjfFd8j6QD7mOs/CX+Kl1JtcI@th1>

Supported: 100rel,histinfo,norefersub,uui

CSeq: 30839111 INVITE

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, UPDATE

P-Asserted-Identity: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>

P-Called-Party-ID: <sip:+XXXXXXXXXX210@sip-trunk.server.com;user=phone>

Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 1177

v=0

o=- 0 0 IN IP4 217.XX.XX.229

s=on transit

c=IN IP4 217.XX.XX.242

t=0 0

m=audio 25428 RTP/SAVP 96 97 9 98 8 99 100 101 102 103 104

b=AS:80

a=rtpmap:96 AMR-WB/16000

a=fmtp:96 mode-set=0,1,2; mode-change-period=2; mode-change-neighbor=1; max-red=0

a=rtpmap:97 AMR-WB/16000

a=fmtp:97 mode-change-capability=2; max-red=0

a=rtpmap:9 G722/8000

a=rtpmap:98 AMR/8000

a=fmtp:98 mode-set=7; max-red=0

a=rtpmap:8 PCMA/8000

a=rtpmap:99 AMR/8000

a=fmtp:99 mode-set=0,1,2,3,4,5,6,7; mode-change-period=2; mode-change-neighbor=1; max-red=0

a=rtpmap:100 AMR/8000

a=fmtp:100 mode-set=0,2,4,7; mode-change-period=2; mode-change-neighbor=1; max-red=0

a=rtpmap:101 AMR/8000

a=fmtp:101 mode-set=0,1,2,3,4,5,6,7; max-red=0

a=rtpmap:102 AMR/8000

a=fmtp:102 mode-set=0,2,4; mode-change-period=2; mode-change-neighbor=1; max-red=0

a=rtpmap:103 telephone-event/8000

a=rtpmap:104 telephone-event/16000

a=ptime:20

a=maxptime:30

a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:DxZ7OV9m5h/8f0yXVZLm/4GY6Vlz644QDhGuXf+4

a=crypto:6 AES_CM_128_HMAC_SHA1_32 inline:wEKyLse50jawQNyWTgDzXYPoYjkGYzggUba2Hb6D

a=crypto:7 F8_128_HMAC_SHA1_80 inline:VYticaw6aZDkijMy784BJpWyDEq79o1FLPj+NtrW

== Setting global variable 'SIPDOMAIN' to '7X.XX.XX.156'

<--- Transmitting SIP response (447 bytes) to TLS:217.XX.XX.229:5061 --->

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 217.XX.XX.229:5061;rport=5061;received=217.XX.XX.229;branch=z9hG4bK033a350245697d3777b63fa6cdd876fd.3acdd739

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

To: <sip:+XXXXXXXXXX210@server.com;user=phone>

CSeq: 30839111 INVITE

Server: Asterisk PBX 17.3.0

Content-Length: 0

== Using SIP RTP Audio TOS bits 184

-- Executing [+XXXXXXXXXX210@incoming-calls:1] Goto("PJSIP/mytrunk-in-00000000", "dialplan,6001,1") in new stack

-- Goto (dialplan,6001,1)

-- Executing [6001@dialplan:1] Dial("PJSIP/mytrunk-in-00000000", "PJSIP/Gilles,20") in new stack

-- Called PJSIP/Gilles

<--- Transmitting SIP request (968 bytes) to UDP:192.168.0.181:30614 --->

INVITE sip:Gilles@192.168.0.181:30614;rinstance=2ce26fd329982ad0 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.190:5060;rport;branch=z9hG4bKPj2f72a4de-3721-4253-81e9-5b7b53d6fc0b

From: <sip:+XXXXXXXXXX395@192.168.0.190>;tag=f7049f85-156d-435f-ae8e-2e5e17832fcd

To: <sip:Gilles@192.168.0.181;rinstance=2ce26fd329982ad0>

Contact: <sip:+XXXXXXXXXX210@192.168.0.190:5060>

Call-ID: 3524acd2-ea5c-4a9a-853e-a1006e1e217e

CSeq: 7949 INVITE

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER

Supported: 100rel, timer, replaces, norefersub

Session-Expires: 1800

Min-SE: 90

Max-Forwards: 70

User-Agent: Asterisk PBX 17.3.0

Content-Type: application/sdp

Content-Length: 239

v=0

o=- 1590928905 1590928905 IN IP4 192.168.0.190

s=Asterisk

c=IN IP4 192.168.0.190

t=0 0

m=audio 15224 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv

<--- Received SIP response (350 bytes) from UDP:192.168.0.181:30614 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.0.190:5060;rport=5060;branch=z9hG4bKPj2f72a4de-3721-4253-81e9-5b7b53d6fc0b

To: <sip:Gilles@192.168.0.181;rinstance=2ce26fd329982ad0>

From: <sip:+XXXXXXXXXX395@192.168.0.190>;tag=f7049f85-156d-435f-ae8e-2e5e17832fcd

Call-ID: 3524acd2-ea5c-4a9a-853e-a1006e1e217e

CSeq: 7949 INVITE

Content-Length: 0

<--- Received SIP response (436 bytes) from UDP:192.168.0.181:30614 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.0.190:5060;rport=5060;branch=z9hG4bKPj2f72a4de-3721-4253-81e9-5b7b53d6fc0b

Contact: <sip:Gilles@192.168.0.181:30614>

To: <sip:Gilles@192.168.0.181;rinstance=2ce26fd329982ad0>;tag=db85ae48

From: <sip:+XXXXXXXXXX395@192.168.0.190>;tag=f7049f85-156d-435f-ae8e-2e5e17832fcd

Call-ID: 3524acd2-ea5c-4a9a-853e-a1006e1e217e

CSeq: 7949 INVITE

User-Agent: SessionTalk 6.0

Content-Length: 0

-- PJSIP/Gilles-00000001 is ringing

-- PJSIP/Gilles-00000001 is ringing

<--- Transmitting SIP response (650 bytes) to TLS:217.XX.XX.229:5061 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 217.XX.XX.229:5061;rport=5061;received=217.XX.XX.229;branch=z9hG4bK033a350245697d3777b63fa6cdd876fd.3acdd739

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

CSeq: 30839111 INVITE

Server: Asterisk PBX 17.3.0

Contact: <sip:7X.XX.XX.156:5061;transport=TLS>

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER

Content-Length: 0

<--- Received SIP response (436 bytes) from UDP:192.168.0.181:30614 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.0.190:5060;rport=5060;branch=z9hG4bKPj2f72a4de-3721-4253-81e9-5b7b53d6fc0b

Contact: <sip:Gilles@192.168.0.181:30614>

To: <sip:Gilles@192.168.0.181;rinstance=2ce26fd329982ad0>;tag=db85ae48

From: <sip:+XXXXXXXXXX395@192.168.0.190>;tag=f7049f85-156d-435f-ae8e-2e5e17832fcd

Call-ID: 3524acd2-ea5c-4a9a-853e-a1006e1e217e

CSeq: 7949 INVITE

User-Agent: SessionTalk 6.0

Content-Length: 0

-- PJSIP/Gilles-00000001 is ringing

-- PJSIP/Gilles-00000001 is ringing

<--- Received SIP response (851 bytes) from UDP:192.168.0.181:30614 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.0.190:5060;rport=5060;branch=z9hG4bKPj2f72a4de-3721-4253-81e9-5b7b53d6fc0b

Require: timer

Contact: <sip:Gilles@192.168.0.181:30614>

To: <sip:Gilles@192.168.0.181;rinstance=2ce26fd329982ad0>;tag=db85ae48

From: <sip:+XXXXXXXXXX395@192.168.0.190>;tag=f7049f85-156d-435f-ae8e-2e5e17832fcd

Call-ID: 3524acd2-ea5c-4a9a-853e-a1006e1e217e

CSeq: 7949 INVITE

Session-Expires: 1800;refresher=uac

Min-SE: 90

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE

Content-Type: application/sdp

Supported: path, replaces, timer, norefersub

User-Agent: SessionTalk 6.0

Content-Length: 185

v=0

o=- 0 1 IN IP4 192.168.0.250

s=-

c=IN IP4 192.168.0.181

t=0 0

m=audio 4006 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

-- PJSIP/Gilles-00000001 answered PJSIP/mytrunk-in-00000000

> 0xffff44046db0 -- Strict RTP learning after remote address set to: 192.168.0.181:4006

<--- Transmitting SIP request (431 bytes) to UDP:192.168.0.181:30614 --->

ACK sip:Gilles@192.168.0.181:30614 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.190:5060;rport;branch=z9hG4bKPj0b7827c6-1274-4b83-bf63-3a8bff869b4c

From: <sip:+XXXXXXXXXX395@192.168.0.190>;tag=f7049f85-156d-435f-ae8e-2e5e17832fcd

To: <sip:Gilles@192.168.0.181;rinstance=2ce26fd329982ad0>;tag=db85ae48

Call-ID: 3524acd2-ea5c-4a9a-853e-a1006e1e217e

CSeq: 7949 ACK

Max-Forwards: 70

User-Agent: Asterisk PBX 17.3.0

Content-Length: 0

> 0xffff4402d650 -- Strict RTP learning after remote address set to: 217.XX.XX.242:25428

<--- Transmitting SIP response (1059 bytes) to TLS:217.XX.XX.229:5061 --->

SIP/2.0 200 OK

Via: SIP/2.0/TLS 217.XX.XX.229:5061;rport=5061;received=217.XX.XX.229;branch=z9hG4bK033a350245697d3777b63fa6cdd876fd.3acdd739

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

CSeq: 30839111 INVITE

Server: Asterisk PBX 17.3.0

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER

Contact: <sip:7X.XX.XX.156:5061;transport=TLS>

Supported: 100rel, timer, replaces, norefersub

Content-Type: application/sdp

Content-Length: 332

v=0

o=- 0 2 IN IP4 7X.XX.XX.156

s=Asterisk

c=IN IP4 7X.XX.XX.156

t=0 0

m=audio 18496 RTP/SAVP 9 8 103

a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:JugglcmsX6xaY3wK3sDS03nvdeyCJLOC9S3a0Xh7

a=rtpmap:9 G722/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:103 telephone-event/8000

a=fmtp:103 0-16

a=ptime:20

a=maxptime:150

a=sendrecv

-- Channel PJSIP/Gilles-00000001 joined 'simple_bridge' basic-bridge <3bacf03c-7897-42fc-80d8-f1f39fd61cd7>

-- Channel PJSIP/mytrunk-in-00000000 joined 'simple_bridge' basic-bridge <3bacf03c-7897-42fc-80d8-f1f39fd61cd7>

[May 18 13:40:59] WARNING[31542][C-00000001]: res_rtp_asterisk.c:7371 ast_rtp_read: RTP Read too short

> 0xffff44046db0 -- Strict RTP switching to RTP target address 192.168.0.181:4006 as source

<--- Received SIP request (632 bytes) from TLS:217.XX.XX.229:5061 --->

ACK sip:7X.XX.XX.156:5061;transport=TLS SIP/2.0

Via: SIP/2.0/TLS 217.XX.XX.229:5061;branch=z9hG4bKbf0c6e7bf650b7d46b5b05bed785ba10.26de2757

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Max-Forwards: 51

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

Contact: <sip:PUOVKRq3x3KGmbVkZYHoGYG4b7dcwIDh7cLE/horRQvFjfFd8j6QD7mOs/CX+Kl1JtcI@th1>

CSeq: 30839111 ACK

Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE

Content-Length: 0

<--- Received SIP request (1166 bytes) from TLS:217.XX.XX.229:5061 --->

INVITE sip:7X.XX.XX.156:5061;transport=TLS SIP/2.0

Via: SIP/2.0/TLS 217.XX.XX.229:5061;branch=z9hG4bK6e18aa4ce2f466ae6ffc38ddfa8b7aad.50943e4d

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Max-Forwards: 51

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

Contact: <sip:PUOVKRq3x3KGmbVkZYHoGYG4b7dcwIDh7cLE/horRQvFjfFd8j6QD7mOs/CX+Kl1JtcI@th1>

CSeq: 30839112 INVITE

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, UPDATE

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 450

v=0

o=- 0 1 IN IP4 217.XX.XX.229

s=on transit

c=IN IP4 217.XX.XX.242

t=0 0

m=audio 25428 RTP/SAVP 9 103

b=AS:80

a=rtpmap:9 G722/8000

a=rtpmap:103 telephone-event/8000

a=ptime:20

a=maxptime:30

a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:DxZ7OV9m5h/8f0yXVZLm/4GY6Vlz644QDhGuXf+4

a=crypto:6 AES_CM_128_HMAC_SHA1_32 inline:wEKyLse50jawQNyWTgDzXYPoYjkGYzggUba2Hb6D

a=crypto:7 F8_128_HMAC_SHA1_80 inline:VYticaw6aZDkijMy784BJpWyDEq79o1FLPj+NtrW

<--- Transmitting SIP response (520 bytes) to TLS:217.XX.XX.229:5061 --->

SIP/2.0 491 Another INVITE transaction in progress

Via: SIP/2.0/TLS 217.XX.XX.229:5061;rport=5061;received=217.XX.XX.229;branch=z9hG4bK6e18aa4ce2f466ae6ffc38ddfa8b7aad.50943e4d

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

CSeq: 30839112 INVITE

Server: Asterisk PBX 17.3.0

Content-Length: 0

<--- Received SIP request (407 bytes) from TLS:217.XX.XX.229:5061 --->

ACK sip:7X.XX.XX.156:5061;transport=TLS SIP/2.0

Via: SIP/2.0/TLS 217.XX.XX.229:5061;branch=z9hG4bK6e18aa4ce2f466ae6ffc38ddfa8b7aad.50943e4d

Max-Forwards: 51

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

CSeq: 30839112 ACK

Content-Length: 0

> 0xffff4402d650 -- Strict RTP switching to RTP target address 217.XX.XX.242:25428 as source

<--- Transmitting SIP request (1183 bytes) to TLS:217.0.26.165:5061 --->

INVITE sip:PUOVKRq3x3KGmbVkZYHoGYG4b7dcwIDh7cLE/horRQvFjfFd8j6QD7mOs/CX+Kl1JtcI@th1 SIP/2.0

Via: SIP/2.0/TLS 7X.XX.XX.156:5061;rport;branch=z9hG4bKPjf54ca2d0-9eab-4ccd-9531-f24ef76fc424;alias

From: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

To: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Contact: <sip:7X.XX.XX.156:5061;transport=TLS>

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

CSeq: 22063 INVITE

Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER

Supported: 100rel, timer, replaces, norefersub

Session-Expires: 1800

Min-SE: 90

Max-Forwards: 70

User-Agent: Asterisk PBX 17.3.0

Content-Type: application/sdp

Content-Length: 356

v=0

o=- 0 3 IN IP4 7X.XX.XX.156

s=Asterisk

c=IN IP4 7X.XX.XX.156

t=0 0

m=audio 18496 RTP/SAVP 0 9 8 103

a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:JugglcmsX6xaY3wK3sDS03nvdeyCJLOC9S3a0Xh7

a=rtpmap:0 PCMU/8000

a=rtpmap:9 G722/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:103 telephone-event/8000

a=fmtp:103 0-16

a=ptime:20

a=maxptime:150

a=sendrecv

<--- Received SIP response (368 bytes) from TLS:217.0.26.165:5061 --->

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 7X.XX.XX.156:5061;rport;branch=z9hG4bKPjf54ca2d0-9eab-4ccd-9531-f24ef76fc424;alias

To: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

From: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

CSeq: 22063 INVITE

Content-Length: 0

<--- Received SIP response (404 bytes) from TLS:217.0.26.165:5061 --->

SIP/2.0 403 Forbidden (R403_REQUEST_NOT_ALLOWED)

Via: SIP/2.0/TLS 7X.XX.XX.156:5061;rport=60399;branch=z9hG4bKPjf54ca2d0-9eab-4ccd-9531-f24ef76fc424;alias

To: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

From: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

CSeq: 22063 INVITE

Content-Length: 0

<--- Transmitting SIP request (543 bytes) to TLS:217.0.26.165:5061 --->

ACK sip:PUOVKRq3x3KGmbVkZYHoGYG4b7dcwIDh7cLE/horRQvFjfFd8j6QD7mOs/CX+Kl1JtcI@th1 SIP/2.0

Via: SIP/2.0/TLS 7X.XX.XX.156:5061;rport;branch=z9hG4bKPjf54ca2d0-9eab-4ccd-9531-f24ef76fc424;alias

From: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

To: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

CSeq: 22063 ACK

Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Max-Forwards: 70

User-Agent: Asterisk PBX 17.3.0

Content-Length: 0

<--- Transmitting SIP request (567 bytes) to TLS:217.0.26.165:5061 --->

BYE sip:PUOVKRq3x3KGmbVkZYHoGYG4b7dcwIDh7cLE/horRQvFjfFd8j6QD7mOs/CX+Kl1JtcI@th1 SIP/2.0

Via: SIP/2.0/TLS 7X.XX.XX.156:5061;rport;branch=z9hG4bKPj8e4f3397-a0b7-495f-9b05-0b0037acd9e7;alias

From: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

To: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

CSeq: 22064 BYE

Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Reason: Q.850;cause=16

Max-Forwards: 70

User-Agent: Asterisk PBX 17.3.0

Content-Length: 0

<--- Received SIP response (401 bytes) from TLS:217.0.26.165:5061 --->

SIP/2.0 403 Forbidden (R403_REQUEST_NOT_ALLOWED)

Via: SIP/2.0/TLS 7X.XX.XX.156:5061;rport=60399;branch=z9hG4bKPj8e4f3397-a0b7-495f-9b05-0b0037acd9e7;alias

To: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

From: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

CSeq: 22064 BYE

Content-Length: 0

-- Channel PJSIP/mytrunk-in-00000000 left 'simple_bridge' basic-bridge <3bacf03c-7897-42fc-80d8-f1f39fd61cd7>

-- Channel PJSIP/Gilles-00000001 left 'simple_bridge' basic-bridge <3bacf03c-7897-42fc-80d8-f1f39fd61cd7>

== Spawn extension (dialplan, 6001, 1) exited non-zero on 'PJSIP/mytrunk-in-00000000'

<--- Transmitting SIP request (455 bytes) to UDP:192.168.0.181:30614 --->

BYE sip:Gilles@192.168.0.181:30614 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.190:5060;rport;branch=z9hG4bKPj37d7b605-672c-4d2e-984d-307072504b34

From: <sip:+XXXXXXXXXX395@192.168.0.190>;tag=f7049f85-156d-435f-ae8e-2e5e17832fcd

To: <sip:Gilles@192.168.0.181;rinstance=2ce26fd329982ad0>;tag=db85ae48

Call-ID: 3524acd2-ea5c-4a9a-853e-a1006e1e217e

CSeq: 7950 BYE

Reason: Q.850;cause=16

Max-Forwards: 70

User-Agent: Asterisk PBX 17.3.0

Content-Length: 0

<--- Received SIP response (428 bytes) from UDP:192.168.0.181:30614 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.0.190:5060;rport=5060;branch=z9hG4bKPj37d7b605-672c-4d2e-984d-307072504b34

Contact: <sip:Gilles@192.168.0.181:30614>

To: <sip:Gilles@192.168.0.181;rinstance=2ce26fd329982ad0>;tag=db85ae48

From: <sip:+XXXXXXXXXX395@192.168.0.190>;tag=f7049f85-156d-435f-ae8e-2e5e17832fcd

Call-ID: 3524acd2-ea5c-4a9a-853e-a1006e1e217e

CSeq: 7950 BYE

User-Agent: SessionTalk 6.0

Content-Length: 0

<--- Received SIP request (1166 bytes) from TLS:217.XX.XX.229:5061 --->

INVITE sip:7X.XX.XX.156:5061;transport=TLS SIP/2.0

Via: SIP/2.0/TLS 217.XX.XX.229:5061;branch=z9hG4bKee9bb324ddd997ee613b51b8bfff50d7.e49343fe

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Max-Forwards: 51

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

Contact: <sip:PUOVKRq3x3KGmbVkZYHoGYG4b7dcwIDh7cLE/horRQvFjfFd8j6QD7mOs/CX+Kl1JtcI@th1>

CSeq: 30839113 INVITE

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, UPDATE

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 450

v=0

o=- 0 2 IN IP4 217.XX.XX.229

s=on transit

c=IN IP4 217.XX.XX.242

t=0 0

m=audio 25428 RTP/SAVP 9 103

b=AS:80

a=rtpmap:9 G722/8000

a=rtpmap:103 telephone-event/8000

a=ptime:20

a=maxptime:30

a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:DxZ7OV9m5h/8f0yXVZLm/4GY6Vlz644QDhGuXf+4

a=crypto:6 AES_CM_128_HMAC_SHA1_32 inline:wEKyLse50jawQNyWTgDzXYPoYjkGYzggUba2Hb6D

a=crypto:7 F8_128_HMAC_SHA1_80 inline:VYticaw6aZDkijMy784BJpWyDEq79o1FLPj+NtrW

<--- Transmitting SIP response (513 bytes) to TLS:217.XX.XX.229:5061 --->

SIP/2.0 481 Call/Transaction Does Not Exist

Via: SIP/2.0/TLS 217.XX.XX.229:5061;rport=5061;received=217.XX.XX.229;branch=z9hG4bKee9bb324ddd997ee613b51b8bfff50d7.e49343fe

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

CSeq: 30839113 INVITE

Server: Asterisk PBX 17.3.0

Content-Length: 0

<--- Received SIP request (407 bytes) from TLS:217.XX.XX.229:5061 --->

ACK sip:7X.XX.XX.156:5061;transport=TLS SIP/2.0

Via: SIP/2.0/TLS 217.XX.XX.229:5061;branch=z9hG4bKee9bb324ddd997ee613b51b8bfff50d7.e49343fe

Max-Forwards: 51

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

CSeq: 30839113 ACK

Content-Length: 0

<--- Received SIP request (693 bytes) from TLS:217.XX.XX.229:5061 --->

BYE sip:7X.XX.XX.156:5061;transport=TLS SIP/2.0

Via: SIP/2.0/TLS 217.XX.XX.229:5061;branch=z9hG4bK699fc673dff50ee1b1f3c24f8212296b.8a5f2a31

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Max-Forwards: 68

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

Contact: <sip:PUOVKRq3x3KGmbVkZYHoGYG4b7dcwIDh7cLE/horRQvFjfFd8j6QD7mOs/CX+Kl1JtcI@th1>

CSeq: 30839114 BYE

Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE

Reason: TSSI;cause=0;text="Call/Transaction Does Not Exist"

Content-Length: 0

<--- Transmitting SIP response (499 bytes) to TLS:217.XX.XX.229:5061 --->

SIP/2.0 481 Call/Transaction Does Not Exist

Via: SIP/2.0/TLS 217.XX.XX.229:5061;received=217.XX.XX.229;branch=z9hG4bK699fc673dff50ee1b1f3c24f8212296b.8a5f2a31

Record-Route: <sip:reg.sip-trunk.server.com;transport=tls;lr>

Call-ID: 765d083a29cfd9de@8X.XX.XX.166

From: <sip:+XXXXXXXXXX395@sip-trunk.server.com;user=phone>;tag=2430f1d6

To: <sip:+XXXXXXXXXX210@server.com;user=phone>;tag=5f20d626-4b77-419a-be9d-3d13363a7dd2

CSeq: 30839114 BYE

Server: Asterisk PBX 17.3.0

Content-Length: 0

There is a re-INVITE clash and things are falling apart as a result. There is an issue[1] already filed for this re-INVITE from Asterisk itself. There is now a patch attached which may resolve the issue.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-28871

Just applied the patch and my incoming call is now working fine - Thanks again.
I still have an issue with my outgoing call where after accepting the call, it is closed automatically after few seconds due to a “480 Temporarily Unavailable” followed by:
– No one is available to answer at this time (1:0/0/0)
– Auto fallthrough, channel ‘PJSIP/Gilles-00000010’ status is ‘NOANSWER’ .

I will create another thread to avoid to mix 2 issues
BR