Hi @spady7,
I replicated the issue and the logs are pasted below:
smcpagtprlv310011*CLI> pjsip set logger verbose on
PJSIP Logging to verbose has been enabled
<--- Received SIP request (2076 bytes) from TLS:52.114.148.0:29381 --->
INVITE sip:+103226398775@domain:5061;user=phone;transport=tls SIP/2.0
FROM: "Hisham Ahmad Jan"<sip:+10000000000;ext=4038@sip.pstnhub.microsoft.com:5061;user=phone>;tag=10395f168fec4692b1e4f685412f328d
TO: <sip:+103226398775@domain:5061;user=phone>
CSEQ: 1 INVITE
CALL-ID: 8fad891962f159da9064b52bcea17f51
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 52.114.148.0:5061;branch=z9hG4bK514034db
RECORD-ROUTE: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>
CONTACT: <sip:api-du-a-jaea.pstnhub.microsoft.com:443;x-i=8a2869a7-c958-4120-b13d-4ab93a71a257;x-c=8fad891962f159da9064b52bcea17f51/d/8/e5782412e3af4af0932fc68d48f71417>
CONTENT-LENGTH: 1102
MIN-SE: 300
SUPPORTED: histinfo,timer
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2024.4.11.1 i.USWE2.14
CONTENT-TYPE: application/sdp
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
P-ASSERTED-IDENTITY: <tel:+10000000000;ext=4038>,<sip:hjan@domain>
PRIVACY: id
SESSION-EXPIRES: 3600
v=0
o=- 11149 0 IN IP4 127.0.0.1
s=session
c=IN IP4 52.115.160.30
b=CT:10000000
t=0 0
m=audio 51484 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118
c=IN IP4 52.115.160.30
a=rtcp:51485
a=ice-ufrag:Xq31
a=ice-pwd:5BaaDWLAqEvflN+mNiBgebHK
a=rtcp-mux
a=candidate:1 1 UDP 2130706431 52.115.160.30 51484 typ srflx raddr 10.0.21.119 rport 51484
a=candidate:1 2 UDP 2130705918 52.115.160.30 51485 typ srflx raddr 10.0.21.119 rport 51485
a=candidate:2 1 tcp-act 2121006078 52.115.160.30 49152 typ srflx raddr 10.0.21.119 rport 49152
a=candidate:2 2 tcp-act 2121006078 52.115.160.30 49152 typ srflx raddr 10.0.21.119 rport 49152
a=label:main-audio
a=mid:1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:epnanrVB8vYAlz7XxwhstrYVY8XivmxlVCwTMymJ|2^31
a=sendrecv
a=rtpmap:104 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:103 SILK/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=ptime:20
<--- Transmitting SIP response (484 bytes) to TLS:52.114.148.0:29381 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 52.114.148.0:5061;rport=29381;received=52.114.148.0;branch=z9hG4bK514034db
Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>
Call-ID: 8fad891962f159da9064b52bcea17f51
From: "Hisham Ahmad Jan" <sip:+10000000000;ext=4038@sip.pstnhub.microsoft.com;user=phone>;tag=10395f168fec4692b1e4f685412f328d
To: <sip:+103226398775@domain;user=phone>
CSeq: 1 INVITE
Server: Asterisk PBX 18.9.0
Content-Length: 0
== DTLS ECDH initialized (automatic), faster PFS enabled
-- Executing [+103226398775@teams:1] NoOp("PJSIP/teams-in-00000009", "PSTN") in new stack
-- Executing [+103226398775@teams:2] Set("PJSIP/teams-in-00000009", "PJSIP_HEADER(add,CALLERID(ALL))= <sip:+10000000000>") in new stack
-- Executing [+103226398775@teams:3] NoOp("PJSIP/teams-in-00000009", "+103226398775") in new stack
-- Executing [+103226398775@teams:4] NoOp("PJSIP/teams-in-00000009", ""Hisham Ahmad Jan" <+10000000000>") in new stack
-- Executing [+103226398775@teams:5] Answer("PJSIP/teams-in-00000009", "") in new stack
> 0x7f5c6c0c1280 -- Strict RTP learning after remote address set to: 52.115.160.30:51484
<--- Transmitting SIP response (1540 bytes) to TLS:52.114.148.0:29381 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.148.0:5061;rport=29381;received=52.114.148.0;branch=z9hG4bK514034db
Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>
Call-ID: 8fad891962f159da9064b52bcea17f51
From: "Hisham Ahmad Jan" <sip:+10000000000;ext=4038@sip.pstnhub.microsoft.com;user=phone>;tag=10395f168fec4692b1e4f685412f328d
To: <sip:+103226398775@domain;user=phone>;tag=1dcba4ba-fc4f-4e50-a447-5a51b739dec7
CSeq: 1 INVITE
Server: Asterisk PBX 18.9.0
Contact: <sip:domain:5061;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 3600;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 718
v=0
o=- 11149 2 IN IP4 199.96.222.145
s=Asterisk
c=IN IP4 199.96.222.145
t=0 0
a=msid-semantic:WMS *
m=audio 27564 RTP/SAVP 0 8 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 CF:51:CB:20:6C:4B:03:2F:82:E3:96:CB:6A:FB:31:9F:36:5C:3D:65:5A:15:60:23:D2:FC:D6:34:DB:DB:0F:2D
a=ice-ufrag:3b05fb7f3fd02c1d00bd465113a4bb26
a=ice-pwd:00fb1931674f5dcf462429f024c96f82
a=candidate:Hc760de91 1 UDP 2130706431 199.96.222.145 27564 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=msid:2eb7aef4-21b4-4dfc-b739-b8e243e4a1db 8511326c-d825-4d34-be40-4cea8575ec6e
a=rtcp-fb:* transport-cc
a=mid:1
<--- Received SIP request (720 bytes) from TLS:52.114.148.0:29381 --->
ACK sip:domain:5061;transport=TLS SIP/2.0
FROM: "Hisham Ahmad Jan"<sip:+10000000000;ext=4038@sip.pstnhub.microsoft.com:5061;user=phone>;tag=10395f168fec4692b1e4f685412f328d
TO: <sip:+103226398775@domain>;user=phone;tag=1dcba4ba-fc4f-4e50-a447-5a51b739dec7
CSEQ: 1 ACK
CALL-ID: 8fad891962f159da9064b52bcea17f51
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 52.114.148.0:5061;branch=z9hG4bK31c48f2d
CONTACT: <sip:api-du-a-jaea.pstnhub.microsoft.com:443;x-i=8a2869a7-c958-4120-b13d-4ab93a71a257;x-c=8fad891962f159da9064b52bcea17f51/d/8/e5782412e3af4af0932fc68d48f71417>
CONTENT-LENGTH: 0
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2024.4.11.1 i.USWE2.14
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
> 0x7f5c6c0c1280 -- Strict RTP learning after ICE completion
<--- Received SIP request (996 bytes) from TLS:52.114.148.0:29390 --->
BYE sip:domain:5061;transport=TLS SIP/2.0
FROM: "Hisham Ahmad Jan"<sip:+10000000000;ext=4038@sip.pstnhub.microsoft.com:5061;user=phone>;tag=10395f168fec4692b1e4f685412f328d
TO: <sip:+103226398775@domain>;user=phone;tag=1dcba4ba-fc4f-4e50-a447-5a51b739dec7
CSEQ: 2 BYE
CALL-ID: 8fad891962f159da9064b52bcea17f51
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 52.114.148.0:5061;branch=z9hG4bKd8f8996f
REASON: Q.850;cause=79;text="8a2869a7-c958-4120-b13d-4ab93a71a257;InternalDiagCode: UnrecognizedTransportProfile, InternalErrorPhrase: Transport profile is not recognized or not compatible"
CONTACT: <sip:api-du-a-jaea.pstnhub.microsoft.com:443;x-i=8a2869a7-c958-4120-b13d-4ab93a71a257;x-c=8fad891962f159da9064b52bcea17f51/d/8/e5782412e3af4af0932fc68d48f71417>
CONTENT-LENGTH: 0
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2024.4.11.1 i.USWE2.14
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
P-ASSERTED-IDENTITY: <tel:+10000000000;ext=4038>,<sip:hjan@domain>
PRIVACY: id
<--- Transmitting SIP response (441 bytes) to TLS:52.114.148.0:29390 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.148.0:5061;rport=29390;received=52.114.148.0;branch=z9hG4bKd8f8996f
Call-ID: 8fad891962f159da9064b52bcea17f51
From: "Hisham Ahmad Jan" <sip:+10000000000;ext=4038@sip.pstnhub.microsoft.com;user=phone>;tag=10395f168fec4692b1e4f685412f328d
To: <sip:+103226398775@domain>;tag=1dcba4ba-fc4f-4e50-a447-5a51b739dec7;user=phone
CSeq: 2 BYE
Server: Asterisk PBX 18.9.0
Content-Length: 0
== Spawn extension (teams, +103226398775, 5) exited non-zero on 'PJSIP/teams-in-00000009'
<--- Received SIP request (536 bytes) from UDP:192.168.70.111:5060 --->
OPTIONS sip:199.96.222.145 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.111:5060;branch=z9hG4bK5d332c4a;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.70.111>;tag=as63fdff7d
To: <sip:199.96.222.145>
Contact: <sip:asterisk@192.168.70.111:5060>
Call-ID: 53ab483b256649086df3ac250adb5240@192.168.70.111:5060
CSeq: 102 OPTIONS
User-Agent: I2c-PBX
Date: Sun, 21 Apr 2024 13:38:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- Transmitting SIP response (839 bytes) to UDP:192.168.70.111:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.70.111:5060;rport=5060;received=192.168.70.111;branch=z9hG4bK5d332c4a
Call-ID: 53ab483b256649086df3ac250adb5240@192.168.70.111:5060
From: "asterisk" <sip:asterisk@192.168.70.111>;tag=as63fdff7d
To: <sip:199.96.222.145>;tag=z9hG4bK5d332c4a
CSeq: 102 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 18.9.0
Content-Length: 0
<--- Transmitting SIP request (482 bytes) to TLS:52.114.132.46:5061 --->
OPTIONS sip:sip.pstnhub.microsoft.com SIP/2.0
Via: SIP/2.0/TLS domain:5061;rport;branch=z9hG4bKPj6c934cc6-4258-4a2f-a62c-5f816b54fb67;alias
From: <sip:teams-out@domain>;tag=6c5b84f4-764c-4360-be02-dfa561955b71
To: <sip:sip.pstnhub.microsoft.com>
Contact: <sip:teams-out@domain:5061;transport=TLS>
Call-ID: 43e05b88-9192-433c-be0a-05d8ac7bafda
CSeq: 35163 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.9.0
Content-Length: 0
<--- Received SIP response (433 bytes) from TLS:52.114.132.46:5061 --->
SIP/2.0 200 OK
FROM: <sip:teams-out@domain>;tag=6c5b84f4-764c-4360-be02-dfa561955b71
TO: <sip:sip.pstnhub.microsoft.com>
CSEQ: 35163 OPTIONS
CALL-ID: 43e05b88-9192-433c-be0a-05d8ac7bafda
VIA: SIP/2.0/TLS domain:5061;branch=z9hG4bKPj6c934cc6-4258-4a2f-a62c-5f816b54fb67;rport
CONTENT-LENGTH: 0
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
SERVER: Microsoft.PSTNHub.SIPProxy v.2024.4.11.1 i.USEA.13
<--- Received SIP request (531 bytes) from TLS:52.114.132.46:23507 --->
OPTIONS sip:teams-out@domain:5061;transport=TLS SIP/2.0
FROM: <sip:sip-du-a-us.pstnhub.microsoft.com:5061>;tag=b8a1792c-7792-4d97-abef-85e56d0b1372
TO: <sip:teams-out@domain>
CSEQ: 1 OPTIONS
CALL-ID: 3db35368-4d6a-46e8-a48d-ad37b4a47fa3
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK6519ff43
CONTACT: <sip:sip-du-a-us.pstnhub.microsoft.com:5061>
CONTENT-LENGTH: 0
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2024.4.11.1 i.USEA.13
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
<--- Transmitting SIP response (860 bytes) to TLS:52.114.132.46:23507 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.132.46:5061;rport=23507;received=52.114.132.46;branch=z9hG4bK6519ff43
Call-ID: 3db35368-4d6a-46e8-a48d-ad37b4a47fa3
From: <sip:sip-du-a-us.pstnhub.microsoft.com>;tag=b8a1792c-7792-4d97-abef-85e56d0b1372
To: <sip:teams-out@domain>;tag=z9hG4bK6519ff43
CSeq: 1 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 18.9.0
Content-Length: 0
I have redacted some info such as the domain name and my office UAN.
Hope this helps!
Best,
Hisham