I am getting an “everyone is busy” dialing from my PJSIP server to a remote chan_sip server.
I would be grateful for any tips and hope this is a complete capture that exposes the problem:
<--- Received SIP request (1171 bytes) from UDP:softclientPrvIP:5060 --->
INVITE sip:RmtPBXExt@AsteriskPrvIPaddr:5060 SIP/2.0
Via: SIP/2.0/UDP softclientPrvIP:5060;branch=z9hG4bK.M6idYLMZd;rport
From: <sip:InternalExt@AsteriskPrvIPaddr>;tag=VoI7kLuBc
To: sip:RmtPBXExt@AsteriskPrvIPaddr
CSeq: 20 INVITE
Call-ID: AAYN7xQHYK
Max-Forwards: 70
Supported: replaces, outbound, gruu, path, record-aware
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 548
Contact: <sip:InternalExt@softclientPrvIP;transport=udp>;expires=3599;+org.linphone.specs="lime"
User-Agent: Linphone-Desktop/5.2.6 (tatooine) fedora/40 Qt/5.15.2 LinphoneSDK/5.3.72
v=0
o=InternalExt 3188 127 IN IP4 softclientPrvIP
s=Talk
c=IN IP4 softclientPrvIP
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
a=record:off
m=audio 54948 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp:40063
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
<--- Transmitting SIP response (453 bytes) to UDP:softclientPrvIP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP softclientPrvIP:5060;rport=5060;received=softclientPrvIP;branch=z9hG4bK.M6idYLMZd
Call-ID: AAYN7xQHYK
From: <sip:InternalExt@AsteriskPrvIPaddr>;tag=VoI7kLuBc
To: <sip:RmtPBXExt@AsteriskPrvIPaddr>;tag=z9hG4bK.M6idYLMZd
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1745972819/2b4eed16fb36add3dd6a6b6e9bed45cd",opaque="07258f23262113db",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.4.0
Content-Length: 0
<--- Received SIP request (348 bytes) from UDP:softclientPrvIP:5060 --->
ACK sip:RmtPBXExt@AsteriskPrvIPaddr:5060 SIP/2.0
Via: SIP/2.0/UDP softclientPrvIP:5060;branch=z9hG4bK.M6idYLMZd;rport
Call-ID: AAYN7xQHYK
From: <sip:InternalExt@AsteriskPrvIPaddr>;tag=VoI7kLuBc
To: <sip:RmtPBXExt@AsteriskPrvIPaddr>;tag=z9hG4bK.M6idYLMZd
Contact: <sip:InternalExt@softclientPrvIP;transport=udp>;expires=3599;+org.linphone.specs="lime"
Max-Forwards: 70
CSeq: 20 ACK
<--- Received SIP request (1454 bytes) from UDP:softclientPrvIP:5060 --->
INVITE sip:RmtPBXExt@AsteriskPrvIPaddr:5060 SIP/2.0
Via: SIP/2.0/UDP softclientPrvIP:5060;branch=z9hG4bK.M4iUw~cUZ;rport
From: <sip:InternalExt@AsteriskPrvIPaddr>;tag=VoI7kLuBc
To: sip:RmtPBXExt@AsteriskPrvIPaddr
CSeq: 21 INVITE
Call-ID: AAYN7xQHYK
Max-Forwards: 70
Supported: replaces, outbound, gruu, path, record-aware
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 548
Contact: <sip:InternalExt@softclientPrvIP;transport=udp>;expires=3599;+org.linphone.specs="lime"
User-Agent: Linphone-Desktop/5.2.6 (tatooine) fedora/40 Qt/5.15.2 LinphoneSDK/5.3.72
Authorization: Digest realm="asterisk", nonce="1745972819/2b4eed16fb36add3dd6a6b6e9bed45cd", algorithm=md5, opaque="07258f23262113db", username="InternalExt", uri="sip:RmtPBXExt@AsteriskPrvIPaddr:5060", response="2338ac3ad70ad1eb05a8817e3d5dcfdc", cnonce="9eIw5rYPBXrSQXhc", nc=00000001, qop=auth
v=0
o=InternalExt 3188 127 IN IP4 softclientPrvIP
s=Talk
c=IN IP4 softclientPrvIP
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
a=record:off
m=audio 54948 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp:40063
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
<--- Transmitting SIP response (279 bytes) to UDP:softclientPrvIP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP softclientPrvIP:5060;rport=5060;received=softclientPrvIP;branch=z9hG4bK.M4iUw~cUZ
Call-ID: AAYN7xQHYK
From: <sip:InternalExt@AsteriskPrvIPaddr>;tag=VoI7kLuBc
To: <sip:RmtPBXExt@AsteriskPrvIPaddr>
CSeq: 21 INVITE
Server: Asterisk PBX 18.4.0
Content-Length: 0
-- Executing [RmtPBXExt@my-phone:1] Dial("PJSIP/InternalExt-00000000", "PJSIP/Ast-B-ext@AsteriskPBX-B") in new stack
-- Called PJSIP/Ast-B-ext@AsteriskPBX-B
<--- Transmitting SIP request (925 bytes) to UDP:Asterisk-B-pubIP:6260 --->
INVITE sip:Ast-B-ext@Asterisk-B-pubIP:6260 SIP/2.0
Via: SIP/2.0/UDP AsteriskPubIPaddr:5060;rport;branch=z9hG4bKPja0d473aa-9316-419c-8151-eb58fbbce49b
From: "MIRMAN JEFF" <sip:2135551212@AsteriskPrvIPaddr>;tag=5f99a357-0d8f-4fdf-be6b-c3f3d4e71728
To: <sip:Ast-B-ext@Asterisk-B-pubIP>
Contact: <sip:asterisk@AsteriskPubIPaddr:5060>
Call-ID: 6c447989-39a3-4cf2-a77d-a458f0ab0da2
CSeq: 29261 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Content-Type: application/sdp
Content-Length: 241
v=0
o=- 2070407527 2070407527 IN IP4 AsteriskPubIPaddr
s=Asterisk
c=IN IP4 AsteriskPubIPaddr
t=0 0
m=audio 17826 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 (619 bytes) from UDP:Asterisk-B-pubIP:6260 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP AsteriskPubIPaddr:5060;branch=z9hG4bKPja0d473aa-9316-419c-8151-eb58fbbce49b;received=AsteriskPubIPaddr;rport=5060
From: "MIRMAN JEFF" <sip:2135551212@AsteriskPrvIPaddr>;tag=5f99a357-0d8f-4fdf-be6b-c3f3d4e71728
To: <sip:Ast-B-ext@Asterisk-B-pubIP>;tag=as081151ed
Call-ID: 6c447989-39a3-4cf2-a77d-a458f0ab0da2
CSeq: 29261 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="254f844d"
Content-Length: 0
<--- Transmitting SIP request (410 bytes) to UDP:Asterisk-B-pubIP:6260 --->
ACK sip:Ast-B-ext@Asterisk-B-pubIP:6260 SIP/2.0
Via: SIP/2.0/UDP AsteriskPubIPaddr:5060;rport;branch=z9hG4bKPja0d473aa-9316-419c-8151-eb58fbbce49b
From: "MIRMAN JEFF" <sip:2135551212@AsteriskPrvIPaddr>;tag=5f99a357-0d8f-4fdf-be6b-c3f3d4e71728
To: <sip:Ast-B-ext@Asterisk-B-pubIP>;tag=as081151ed
Call-ID: 6c447989-39a3-4cf2-a77d-a458f0ab0da2
CSeq: 29261 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Content-Length: 0
<--- Transmitting SIP request (1097 bytes) to UDP:Asterisk-B-pubIP:6260 --->
INVITE sip:Ast-B-ext@Asterisk-B-pubIP:6260 SIP/2.0
Via: SIP/2.0/UDP AsteriskPubIPaddr:5060;rport;branch=z9hG4bKPj4273d65d-632f-4837-af98-8f93675d7d2e
From: "MIRMAN JEFF" <sip:2135551212@AsteriskPrvIPaddr>;tag=5f99a357-0d8f-4fdf-be6b-c3f3d4e71728
To: <sip:Ast-B-ext@Asterisk-B-pubIP>
Contact: <sip:asterisk@AsteriskPubIPaddr:5060>
Call-ID: 6c447989-39a3-4cf2-a77d-a458f0ab0da2
CSeq: 29262 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Authorization: Digest username="AsteriskPBX-A", realm="asterisk", nonce="254f844d", uri="sip:Ast-B-ext@Asterisk-B-pubIP:6260", response="dedcdc0ca7ea6609abf9da2e9121533a", algorithm=MD5
Content-Type: application/sdp
Content-Length: 241
v=0
o=- 2070407527 2070407527 IN IP4 AsteriskPubIPaddr
s=Asterisk
c=IN IP4 AsteriskPubIPaddr
t=0 0
m=audio 17826 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 (540 bytes) from UDP:Asterisk-B-pubIP:6260 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP AsteriskPubIPaddr:5060;branch=z9hG4bKPj4273d65d-632f-4837-af98-8f93675d7d2e;received=AsteriskPubIPaddr;rport=5060
From: "MIRMAN JEFF" <sip:2135551212@AsteriskPrvIPaddr>;tag=5f99a357-0d8f-4fdf-be6b-c3f3d4e71728
To: <sip:Ast-B-ext@Asterisk-B-pubIP>;tag=as081151ed
Call-ID: 6c447989-39a3-4cf2-a77d-a458f0ab0da2
CSeq: 29262 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- Transmitting SIP request (410 bytes) to UDP:Asterisk-B-pubIP:6260 --->
ACK sip:Ast-B-ext@Asterisk-B-pubIP:6260 SIP/2.0
Via: SIP/2.0/UDP AsteriskPubIPaddr:5060;rport;branch=z9hG4bKPj4273d65d-632f-4837-af98-8f93675d7d2e
From: "MYNAME" <sip:2135551212@AsteriskPrvIPaddr>;tag=5f99a357-0d8f-4fdf-be6b-c3f3d4e71728
To: <sip:Ast-B-ext@Asterisk-B-pubIP>;tag=as081151ed
Call-ID: 6c447989-39a3-4cf2-a77d-a458f0ab0da2
CSeq: 29262 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Content-Length: 0
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [RmtPBXExt@my-phone:2] Hangup("PJSIP/InternalExt-00000000", "") in new stack
== Spawn extension (my-phone, RmtPBXExt, 2) exited non-zero on 'PJSIP/InternalExt-00000000'
<--- Transmitting SIP response (347 bytes) to UDP:softclientPrvIP:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP softclientPrvIP:5060;rport=5060;received=softclientPrvIP;branch=z9hG4bK.M4iUw~cUZ
Call-ID: AAYN7xQHYK
From: <sip:InternalExt@AsteriskPrvIPaddr>;tag=VoI7kLuBc
To: <sip:RmtPBXExt@AsteriskPrvIPaddr>;tag=fcdbae40-051d-404f-b08a-00002ccfa7bf
CSeq: 21 INVITE
Server: Asterisk PBX 18.4.0
Reason: Q.850;cause=21
Content-Length: 0
<--- Received SIP request (367 bytes) from UDP:softclientPrvIP:5060 --->
ACK sip:RmtPBXExt@AsteriskPrvIPaddr:5060 SIP/2.0
Via: SIP/2.0/UDP softclientPrvIP:5060;branch=z9hG4bK.M4iUw~cUZ;rport
Call-ID: AAYN7xQHYK
From: <sip:InternalExt@AsteriskPrvIPaddr>;tag=VoI7kLuBc
To: <sip:RmtPBXExt@AsteriskPrvIPaddr>;tag=fcdbae40-051d-404f-b08a-00002ccfa7bf
Contact: <sip:InternalExt@softclientPrvIP;transport=udp>;expires=3599;+org.linphone.specs="lime"
Max-Forwards: 70
CSeq: 21 ACK