No media after upgrade to asterisk 22

Asterisk Asterisk SIP
1

In the appended log, the incoming call from sipgate is directed to 83,
which rings and sends back RINGING to sipgate.
At line 262, I see:
– PJSIP/83-00000037 answered PJSIP/sipgate-00000036
<— Transmitting SIP response (1493 bytes) to UDP:[2001:ab7::10]:5060 —>
But no ANSWER is sent to sipgate, so sipgate does not turn on media stream.

83 has G722, PCMU PCMA and G29 enabled.

How can I fix this?
Please advice.
ajr

snippet from pjsip.conf

endpoint_mobile](!)
type = endpoint
context = from_mobile
allow = !all,g722,ulaw
transport = transport_udp_ipv6
rtp_ipv6 = yes
rtp_symmetric = yes
force_rport = yes ; FW
direct_media=no ; FW FIXME to be tested
media_address = [2001:14f8:21c:1f::211]
bind_rtp_to_media_address = yes
trust_id_outbound = yes
dtmf_mode = rfc4733
from_domain = voip.lrau.net
identify_by=username ; TEST: client IP changes
;
; Extension and Device state options
;
;;contacts=1
allow_subscribe=yes
sub_min_expiry=30

auth_userpass_mobile
type = auth
auth_type = userpass

aor_single_mobile
type = aor
max_contacts = 2

…

;================================
; Axels Yealink T53W
;================================

83
auth = 83
aors = 83
callerid = Axel Rau <83>

83
password = some_pw
username = some_username

83
mailboxes = some_email

;================================

Call log:

<— Received SIP request (1555 bytes) from UDP:[2001:ab7::10]:5060 —>
INVITE sip:49699514180@[2001:14f8:21c:1f::211]:5060 SIP/2.0
Record-Route: sip:[2001:ab7::10];r2=on;lr;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;r2=on;lr;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Via: SIP/2.0/UDP [2001:ab7::10];branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de
Contact: sip:01607568212@212.9.44.67:5060
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
CSeq: 32071 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, MESSAGE, REFER
Supported: replaces, histinfo
Max-Forwards: 67
Content-Type: application/sdp
Content-Length: 418

v=0
o=- 133932663 133932663 IN IP6 2001:ab7:3000:2::175
s=sGW
t=0 0
m=audio 25156 RTP/AVP 107 9 8 0 3 101 113
c=IN IP6 2001:ab7:3000:2::175
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=sendrecv
a=rtcp:25157
a=ptime:20
a=maxptime:20

<— Transmitting SIP response (935 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

-- Executing [49699514180@from_sipgate:1] Goto("PJSIP/sipgate-00000036", "DD0,49699514180,1") in new stack
-- Goto (DD0,49699514180,1)
-- Executing [49699514180@DD0:1] Verbose("PJSIP/sipgate-00000036", "1, "DD0: User 01607568212 dialed 49699514180."") in new stack

“DD0: User 01607568212 dialed 49699514180.”
– Executing [49699514180@DD0:2] Progress(“PJSIP/sipgate-00000036”, “”) in new stack
– Executing [49699514180@DD0:3] Dial(“PJSIP/sipgate-00000036”, “PJSIP/83,20”) in new stack

0x23279c8c6000 – Strict RTP learning after remote address set to: [2001:ab7:3000:2::175]:25156
<— Transmitting SIP response (1459 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Contact: sip:[2001:14f8:21c:1f::211]:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 133932663 133932665 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 20614 RTP/AVP 9 0 113
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Transmitting SIP request (1005 bytes) to UDP:[2001:14f8:21c:10::129]:5060 —>
INVITE sip:83@[2001:14f8:21c:10::129]:5060 SIP/2.0
Via: SIP/2.0/UDP [2001:14f8:21c:1f::211]:5060;rport;branch=z9hG4bKPj253aff78-8297-11f0-ae2e-ac1f6bb1c248
From: “01607568212” sip:01607568212@voip.lrau.net;tag=253a68b4-8297-11f0-ae2e-ac1f6bb1c248
To: sip:83@[2001:14f8:21c:10::129]
Contact: sip:asterisk@[2001:14f8:21c:1f::211]:5060
Call-ID: 253a69ff-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 20569 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.5.1
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 843704762 843704762 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 26092 RTP/AVP 9 0 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

-- Called PJSIP/83

<— Received SIP response (391 bytes) from UDP:[2001:14f8:21c:10::129]:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [2001:14f8:21c:1f::211]:5060;rport=5060;branch=z9hG4bKPj253aff78-8297-11f0-ae2e-ac1f6bb1c248
From: “01607568212” sip:01607568212@voip.lrau.net;tag=253a68b4-8297-11f0-ae2e-ac1f6bb1c248
To: sip:83@[2001:14f8:21c:10::129]
Call-ID: 253a69ff-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 20569 INVITE
User-Agent: Yealink SIP-T53W 96.85.0.5
Content-Length: 0

<— Received SIP response (625 bytes) from UDP:[2001:14f8:21c:10::129]:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP [2001:14f8:21c:1f::211]:5060;rport=5060;branch=z9hG4bKPj253aff78-8297-11f0-ae2e-ac1f6bb1c248
From: “01607568212” sip:01607568212@voip.lrau.net;tag=253a68b4-8297-11f0-ae2e-ac1f6bb1c248
To: sip:83@[2001:14f8:21c:10::129];tag=4010040018
Call-ID: 253a69ff-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 20569 INVITE
Contact: sip:83@[2001:14f8:21c:10::129]:5060
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T53W 96.85.0.5
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

-- PJSIP/83-00000037 is ringing

<— Transmitting SIP response (1459 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:[2001:14f8:21c:1f::211]:5060
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 133932663 133932665 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 20614 RTP/AVP 9 0 113
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Received SIP request (437 bytes) from UDP:[2001:ab7::10]:5060 —>
OPTIONS sip:2846812t1@[2001:14f8:21c:1f::211]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP [2001:ab7::10];branch=z9hG4bK5308.6357c7114b1b4848ffa36eff53302db5.0
Via: SIP/2.0/UDP 172.20.40.7:5060;branch=z9hG4bK2480760
From: sip:keepalive@sipgate.de;tag=uloc-68a5d765-3a9d1c-27ada3-32953b42-a9af61a6
To: sip:2846812t1@[2001:14f8:21c:1f::211]:5060
Call-ID: e830a52-0dc3f3b7-a432e27@172.20.40.7
CSeq: 1 OPTIONS
Content-Length: 0

<— Transmitting SIP response (986 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK5308.6357c7114b1b4848ffa36eff53302db5.0
Via: SIP/2.0/UDP 172.20.40.7:5060;branch=z9hG4bK2480760
Call-ID: e830a52-0dc3f3b7-a432e27@172.20.40.7
From: sip:keepalive@sipgate.de;tag=uloc-68a5d765-3a9d1c-27ada3-32953b42-a9af61a6
To: sip:2846812t1@[2001:14f8:21c:1f::211];tag=z9hG4bK5308.6357c7114b1b4848ffa36eff53302db5.0
CSeq: 1 OPTIONS
Accept: application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Received SIP response (903 bytes) from UDP:[2001:14f8:21c:10::129]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:14f8:21c:1f::211]:5060;rport=5060;branch=z9hG4bKPj253aff78-8297-11f0-ae2e-ac1f6bb1c248
From: “01607568212” sip:01607568212@voip.lrau.net;tag=253a68b4-8297-11f0-ae2e-ac1f6bb1c248
To: sip:83@[2001:14f8:21c:10::129];tag=4010040018
Call-ID: 253a69ff-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 20569 INVITE
Contact: sip:83@[2001:14f8:21c:10::129]:5060
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T53W 96.85.0.5
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 229

v=0
o=- 20016 20016 IN IP6 2001:14f8:21c:10::129
s=SDP data
c=IN IP6 2001:14f8:21c:10::129
t=0 0
m=audio 12746 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

   > 0x23279c8c9000 -- Strict RTP learning after remote address set to: [2001:14f8:21c:10::129]:12746

<— Transmitting SIP request (439 bytes) to UDP:[2001:14f8:21c:10::129]:5060 —>
ACK sip:83@[2001:14f8:21c:10::129]:5060 SIP/2.0
Via: SIP/2.0/UDP [2001:14f8:21c:1f::211]:5060;rport;branch=z9hG4bKPj2bd0e71e-8297-11f0-ae2e-ac1f6bb1c248
From: “01607568212” sip:01607568212@voip.lrau.net;tag=253a68b4-8297-11f0-ae2e-ac1f6bb1c248
To: sip:83@[2001:14f8:21c:10::129];tag=4010040018
Call-ID: 253a69ff-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 20569 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.5.1
Content-Length: 0

-- PJSIP/83-00000037 answered PJSIP/sipgate-00000036

<— Transmitting SIP response (1493 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:[2001:14f8:21c:1f::211]:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 133932663 133932665 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 20614 RTP/AVP 9 0 113
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

-- Channel PJSIP/83-00000037 joined 'simple_bridge' basic-bridge <79a9ffdd-84fb-4010-b63d-71a0ee1b2574>
-- Channel PJSIP/sipgate-00000036 joined 'simple_bridge' basic-bridge <79a9ffdd-84fb-4010-b63d-71a0ee1b2574>
   > Bridge 79a9ffdd-84fb-4010-b63d-71a0ee1b2574: switching from simple_bridge technology to native_rtp
   > Locally RTP bridged 'PJSIP/sipgate-00000036' and 'PJSIP/83-00000037' in stack
   > 0x23279c8c9000 -- Strict RTP switching to RTP target address [2001:14f8:21c:10::129]:12746 as source

<— Transmitting SIP response (1493 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:[2001:14f8:21c:1f::211]:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 133932663 133932665 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 20614 RTP/AVP 9 0 113
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Transmitting SIP response (1493 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:[2001:14f8:21c:1f::211]:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 133932663 133932665 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 20614 RTP/AVP 9 0 113
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Transmitting SIP response (1493 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:[2001:14f8:21c:1f::211]:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 133932663 133932665 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 20614 RTP/AVP 9 0 113
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

   > 0x23279c8c9000 -- Strict RTP learning complete - Locking on source address [2001:14f8:21c:10::129]:12746

<— Received SIP request (461 bytes) from UDP:[2001:14f8:21c:10::129]:5060 —>
BYE sip:asterisk@[2001:14f8:21c:1f::211]:5060 SIP/2.0
Via: SIP/2.0/UDP [2001:14f8:21c:10::129]:5060;branch=z9hG4bK1323195091
From: sip:83@[2001:14f8:21c:10::129];tag=4010040018
To: “01607568212” sip:01607568212@voip.lrau.net;tag=253a68b4-8297-11f0-ae2e-ac1f6bb1c248
Call-ID: 253a69ff-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 2 BYE
Contact: sip:83@[2001:14f8:21c:10::129]:5060
Max-Forwards: 70
User-Agent: Yealink SIP-T53W 96.85.0.5
Content-Length: 0

<— Transmitting SIP response (388 bytes) to UDP:[2001:14f8:21c:10::129]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:14f8:21c:10::129]:5060;rport=5060;received=2001:14f8:21c:10::129;branch=z9hG4bK1323195091
Call-ID: 253a69ff-8297-11f0-ae2e-ac1f6bb1c248
From: sip:83@[2001:14f8:21c:10::129];tag=4010040018
To: “01607568212” sip:01607568212@voip.lrau.net;tag=253a68b4-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 2 BYE
Server: Asterisk PBX 22.5.1
Content-Length: 0

-- Channel PJSIP/83-00000037 left 'native_rtp' basic-bridge <79a9ffdd-84fb-4010-b63d-71a0ee1b2574>
-- Channel PJSIP/sipgate-00000036 left 'native_rtp' basic-bridge <79a9ffdd-84fb-4010-b63d-71a0ee1b2574>

== Spawn extension (DD0, 49699514180, 3) exited non-zero on ‘PJSIP/sipgate-00000036’
<— Transmitting SIP response (1493 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:[2001:14f8:21c:1f::211]:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 133932663 133932665 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 20614 RTP/AVP 9 0 113
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Received SIP request (690 bytes) from UDP:[2001:ab7::10]:5060 —>
CANCEL sip:49699514180@[2001:14f8:21c:1f::211]:5060 SIP/2.0
Record-Route: sip:[2001:ab7::10];r2=on;lr;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;r2=on;lr;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Via: SIP/2.0/UDP [2001:ab7::10];branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
CSeq: 32071 CANCEL
Max-Forwards: 67
Content-Length: 0
Reason: Q.850;cause=127

<— Transmitting SIP response (711 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 CANCEL
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Received SIP request (437 bytes) from UDP:[2001:ab7::10]:5060 —>
OPTIONS sip:2846812t1@[2001:14f8:21c:1f::211]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP [2001:ab7::10];branch=z9hG4bKd8b9.cea6d7153f1a92a414de95055d6ad31a.0
Via: SIP/2.0/UDP 172.20.40.7:5060;branch=z9hG4bK8874329
From: sip:keepalive@sipgate.de;tag=uloc-68a5d765-3a9d1c-27ada3-32953b42-ab4971a6
To: sip:2846812t1@[2001:14f8:21c:1f::211]:5060
Call-ID: e830a52-0f6df3b7-8532e27@172.20.40.7
CSeq: 1 OPTIONS
Content-Length: 0

<— Transmitting SIP response (986 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bKd8b9.cea6d7153f1a92a414de95055d6ad31a.0
Via: SIP/2.0/UDP 172.20.40.7:5060;branch=z9hG4bK8874329
Call-ID: e830a52-0f6df3b7-8532e27@172.20.40.7
From: sip:keepalive@sipgate.de;tag=uloc-68a5d765-3a9d1c-27ada3-32953b42-ab4971a6
To: sip:2846812t1@[2001:14f8:21c:1f::211];tag=z9hG4bKd8b9.cea6d7153f1a92a414de95055d6ad31a.0
CSeq: 1 OPTIONS
Accept: application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Transmitting SIP response (1493 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK7a5f.6868dbfcdb0878d3d2c28c93c733d3c6.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK7a5f.3ff1a7e44bb6aad73c74855bd6907be6.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK7a5f.455dfaee17832d0bf066ba07e86cc958.0
Via: SIP/2.0/UDP 212.9.44.67:5060;branch=z9hG4bKPja04d590a-8173-4acb-84a4-fcc488f9a530
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
Record-Route: sip:172.20.40.7;lr;did=1a9.1041
Record-Route: sip:217.10.68.137;lr
Call-ID: b40fb5f1-c12c-43f0-9496-41d2da38e0e1
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=d16ce10a-2cf4-451c-98eb-f5a9b4974c98
To: sip:0049699514180@sipconnect.sipgate.de;tag=253907e6-8297-11f0-ae2e-ac1f6bb1c248
CSeq: 32071 INVITE
Server: Asterisk PBX 22.5.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:[2001:14f8:21c:1f::211]:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 133932663 133932665 IN IP6 2001:14f8:21c:1f::211
s=Asterisk
c=IN IP6 2001:14f8:21c:1f::211
t=0 0
m=audio 20614 RTP/AVP 9 0 113
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:113 telephone-event/8000
a=fmtp:113 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

voice3*CLI>

2

According to that Asterisk is sending a 200 OK for answer, but getting no ACK. I would suggest verifying all the IP addresses and doing a packet capture to examine the actual traffic - including whether an ACK is coming in.

3

It’s not getting any re-transmissions of the INVITE, so the A party seems to have seen some responses, but it is getting a CANCEL, which means they haven’t seen the final response.

The packets are rather long, so my guess is that IP fragmentation is broken. This can happen if firewalls white list on the destination port number, and don’t let through continuation packets. This is most likely to be at the remote end.

Disabling fragmenation locally could also cause problems.

4

I removed the don’t fragment scrub rule on the opnsense firewall. But I see no success.

wg0.txt (25.2 KB)

asterisk.txt (556.5 KB)

I captured a call on both ends of the firwall. Adresses are

  • 2001:14f8:21c:1f::211 asterisk
  • 2001:ab7::10 sipgate
  • 2001:14f8:21c:10::129 IPphone (83)
5

I don’t really have anything else to add beyond my initial analysis. You have to determine why either the remote side is not getting the 200 OK, or why we aren’t getting the ACK.

6

UPDATE:

We identified a fragmentation problem, but could not yet resolve it.

Meanwhile, I found a different problem (or maxbe related?):

On incoming calls, ringing is not propagated to caller (sipgate), instead “Session Progress” is sent:

INVITE sipgate → asterisk
100 trying asterisk → sipgate
183 Session Progress asterisk → sipgate
INVITE asterisk → 83
100 trying 83 → asterisk
180 ringing 83 → asterisk
183 Session Progress asterisk → sipgate

media asterisk → sipgate

Excerpt from full log:

[Aug 28 22:50:01] DEBUG[110225] res_pjsip_session.c: PJSIP/83-00000065: Response is 180 Ringing
[Aug 28 22:50:01] DEBUG[110225] chan_pjsip.c: PJSIP/83-00000065: Status: 180
[Aug 28 22:50:01] DEBUG[110225] chan_pjsip.c: PJSIP/83-00000065: Queueing RINGING
[Aug 28 22:50:01] DEBUG[110225] chan_pjsip.c: PJSIP/83-00000065
[Aug 28 22:50:01] DEBUG[106984] devicestate.c: No provider found, checking channel drivers for PJSIP - 83
[Aug 28 22:50:01] DEBUG[110225] chan_pjsip.c: PJSIP/83-00000065: Status: 180
[Aug 28 22:50:01] VERBOSE[117939][C-00000030] app_dial.c: PJSIP/83-00000065 is ringing
[Aug 28 22:50:01] DEBUG[106984] devicestate.c: Changing state for PJSIP/83 - state 6 (Ringing)
[Aug 28 22:50:01] DEBUG[110225] chan_pjsip.c: PJSIP/83-00000065
[Aug 28 22:50:01] DEBUG[117939][C-00000030] chan_pjsip.c: PJSIP/sipgate-00000064: Indicated Ringing
[Aug 28 22:50:01] DEBUG[110225] res_pjsip_session.c: PJSIP/83-00000065
[Aug 28 22:50:01] DEBUG[110225] res_pjsip_session.c: Nothing delayed
[Aug 28 22:50:01] DEBUG[110225] res_pjsip_session.c: PJSIP/83-00000065 TSX State: Proceeding Inv State: EARLY
[Aug 28 22:50:01] DEBUG[117939][C-00000030] chan_pjsip.c: PJSIP/sipgate-00000064
[Aug 28 22:50:01] DEBUG[107051] app_queue.c: Device ‘PJSIP/83’ changed to state ‘6’ (Ringing) but we don’t care because they’re not a member of any queue.
[Aug 28 22:50:01] DEBUG[110225] res_pjsip_session.c: Topology: Pending: <0:audio-0:audio:sendrecv (g722|ulaw)> Active: (null topology)
[Aug 28 22:50:01] DEBUG[117939][C-00000030] channel.c: Driver for channel ‘PJSIP/sipgate-00000064’ does not support indication 3, emulating it
[Aug 28 22:50:01] DEBUG[110225] res_pjsip_session.c:
[Aug 28 22:50:01] DEBUG[106984] devicestate.c: No provider found, checking channel drivers for PJSIP - sipgate
[Aug 28 22:50:01] DEBUG[117939][C-00000030] channel.c: Channel PJSIP/sipgate-00000064 setting write format path: slin → g722
[Aug 28 22:50:01] DEBUG[117939][C-00000030] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Aug 28 22:50:01] DEBUG[106984] devicestate.c: Changing state for PJSIP/sipgate - state 2 (In use)
[Aug 28 22:50:01] DEBUG[117939][C-00000030] channel.c: Prodding channel ‘PJSIP/sipgate-00000064’
[Aug 28 22:50:01] DEBUG[117877] res_pjsip_session.c: PJSIP/sipgate-00000064: Method is INVITE, Response is 183 Session Progress
[Aug 28 22:50:01] DEBUG[117939][C-00000030] res_rtp_asterisk.c: (0x23279f0af720) RTP received frame with no data for instance, so dropping frame
[Aug 28 22:50:01] DEBUG[117877] res_pjsip_session/pjsip_session_reason_header.c: PJSIP/sipgate-00000064: Response Code: 183
[Aug 28 22:50:01] DEBUG[117877] res_pjsip_session/pjsip_session_reason_header.c: PJSIP/sipgate-00000064: No datastore on session. Nothing to do
[Aug 28 22:50:01] DEBUG[117939][C-00000030] chan_pjsip.c: PJSIP/sipgate-00000064: Indicated Private Cause Code
[Aug 28 22:50:01] DEBUG[117877] res_pjsip_session.c: PJSIP/sipgate-00000064
[Aug 28 22:50:01] DEBUG[117939][C-00000030] chan_pjsip.c: PJSIP/sipgate-00000064
[Aug 28 22:50:01] DEBUG[117877] res_pjsip/pjsip_message_filter.c: Re-wrote Contact URI host/port to 2001:14f8:21c:1f::211:5060 (this may be re-written again later)
[Aug 28 22:50:01] VERBOSE[117877] res_pjsip_logger.c: <— Transmitting SIP response (1460 bytes) to UDP:[2001:ab7::10]:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP [2001:ab7::10];rport=5060;received=2001:ab7::10;branch=z9hG4bK407a.944b7e09b11e295827970895f3847bcb.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK407a.efa8f6d7b014722b740b848df5ddb5d1.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK407a.57d19e63952513750053702d06e7e664.0
Via: SIP/2.0/UDP 217.10.77.72:5060;branch=z9hG4bKPj23eb48b8-48d8-4bbe-9861-d828751242e4
Record-Route: sip:[2001:ab7::10]:5060;lr;r2=on;ftag=da327915-d312-43db-a98f-454bb7d4d34e
Record-Route: sip:217.10.78.108:5064;lr;r2=on;ftag=da327915-d312-43db-a98f-454bb7d4d34e
Record-Route: sip:172.20.40.7;lr;did=04e.ea61
Record-Route: sip:217.10.68.137;lr
Call-ID: 6e75f965-e8a1-4a72-8410-9f41b2be9834
From: “01607568212” sip:01607568212@sipconnect.sipgate.de;tag=da327915-d312-43db-a98f-454bb7d4d34e
To: sip:0049699514180@sipconnect.sipgate.de;tag=910f2efb-8450-11f0-ae2e-ac1f6bb1c248
CSeq: 29232 INVITE

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Log starts too late. It needs to start before the A leg starts, not after the B leg call is requested.

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Additionally if Asterisk has sent a 183 then it will not transition to a 180 afterwards unless the option to do so is enabled[1]. If not enabled then ringing will be sent as in-band audio.

[1] asterisk/configs/samples/pjsip.conf.sample at master · asterisk/asterisk · GitHub

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[RESOLVED]

We finally fixed the framentation issue (at the far end of the wireguard tunnel).

Turning on allow_sending_180_after_183 fixed the other issue.

jcolp and david551 thanks a lot for your help,

ajr

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