OK, here is it. Summary cause I can’t attach a file :
Register from : UDP:176.12.16.218:46212
address in pjsip show contacts : sip:102@176.12.16.218:46212;ob
The fact is that it even does not try to send the invite…
Log follows :
<--- Received SIP request (568 bytes) from UDP:176.12.16.218:46212 --->
REGISTER sip:wrtc.bgtel.bg SIP/2.0
Via: SIP/2.0/UDP 176.12.16.218:46212;rport;branch=z9hG4bKPjJ4e33nW3-E15pBIhp-KNP8RFlRXzJYTP
Route: <sip:wrtc.bgtel.bg;transport=udp;lr>
Max-Forwards: 70
From: <sip:102@wrtc.bgtel.bg>;tag=89TCQmuGCsKfG5BB6W.sVbT2f21jDzng
To: <sip:102@wrtc.bgtel.bg>
Call-ID: 6n9qfcoftZTpt9vBifZ5.mM-dn5Q2qJQ
CSeq: 123 REGISTER
User-Agent: CSipSimple_ali_n-28/r2457
Contact: <sip:102@176.12.16.218:46212;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<--- Transmitting SIP response (551 bytes) to UDP:176.12.16.218:46212 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 176.12.16.218:46212;rport=46212;received=176.12.16.218;branch=z9hG4bKPjJ4e33nW3-E15pBIhp-KNP8RFlRXzJYTP
Call-ID: 6n9qfcoftZTpt9vBifZ5.mM-dn5Q2qJQ
From: <sip:102@wrtc.bgtel.bg>;tag=89TCQmuGCsKfG5BB6W.sVbT2f21jDzng
To: <sip:102@wrtc.bgtel.bg>;tag=z9hG4bKPjJ4e33nW3-E15pBIhp-KNP8RFlRXzJYTP
CSeq: 123 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1559828512/9c010bc1ceb956663f43cc52b5b8f2a7",opaque="1cc3f47c667a3cd6",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.7.2
Content-Length: 0
<--- Received SIP request (855 bytes) from UDP:176.12.16.218:46212 --->
REGISTER sip:wrtc.bgtel.bg SIP/2.0
Via: SIP/2.0/UDP 176.12.16.218:46212;rport;branch=z9hG4bKPjTwBs0JuVvGWXYC1whK1WZwnn5LAv6suO
Route: <sip:wrtc.bgtel.bg;transport=udp;lr>
Max-Forwards: 70
From: <sip:102@wrtc.bgtel.bg>;tag=89TCQmuGCsKfG5BB6W.sVbT2f21jDzng
To: <sip:102@wrtc.bgtel.bg>
Call-ID: 6n9qfcoftZTpt9vBifZ5.mM-dn5Q2qJQ
CSeq: 124 REGISTER
User-Agent: CSipSimple_ali_n-28/r2457
Contact: <sip:102@176.12.16.218:46212;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="102", realm="asterisk", nonce="1559828512/9c010bc1ceb956663f43cc52b5b8f2a7", uri="sip:wrtc.bgtel.bg", response="4c15b5076142ddeaeda97cab683445a5", algorithm=md5, cnonce="26Tw4gbzdyqqNGShuZVrESHiKaCDgBZ6", opaque="42ce852216b66449", qop=auth, nc=00000001
Content-Length: 0
-- Added contact 'sip:102@176.12.16.218:46212;ob' to AOR '102' with expiration of 900 seconds
<--- Transmitting SIP response (501 bytes) to UDP:176.12.16.218:46212 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 176.12.16.218:46212;rport=46212;received=176.12.16.218;branch=z9hG4bKPjTwBs0JuVvGWXYC1whK1WZwnn5LAv6suO
Call-ID: 6n9qfcoftZTpt9vBifZ5.mM-dn5Q2qJQ
From: <sip:102@wrtc.bgtel.bg>;tag=89TCQmuGCsKfG5BB6W.sVbT2f21jDzng
To: <sip:102@wrtc.bgtel.bg>;tag=z9hG4bKPjTwBs0JuVvGWXYC1whK1WZwnn5LAv6suO
CSeq: 124 REGISTER
Date: Thu, 06 Jun 2019 13:41:52 GMT
Contact: <sip:102@176.12.16.218:46212;ob>;expires=899
Expires: 900
Server: Asterisk PBX 15.7.2
Content-Length: 0
...
== Endpoint 102 is now Reachable
-- Contact 102/sip:102@176.12.16.218:46212;ob is now Reachable. RTT: 107.759 msec
vmi214442*CLI> pjsip show contacts
Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================
Contact: 101/sips:101@176.12.16.218:58448;transport=ws; eb17f306df Avail 63.587
Contact: 102/sip:102@176.12.16.218:46212;ob 94e44a37a9 Avail 107.759
Objects found: 2
<--- Transmitting SIP request (431 bytes) to UDP:176.12.16.218:46212 --->
OPTIONS sip:102@176.12.16.218:46212;ob SIP/2.0
Via: SIP/2.0/UDP 207.180.224.10:5060;rport;branch=z9hG4bKPj5c49ef41-e7a5-4a72-bdd3-622302b828aa
From: <sip:102@207.180.224.10>;tag=d30ee158-12c9-49f6-bc84-bf1793c7c29c
To: <sip:102@176.12.16.218;ob>
Contact: <sip:102@207.180.224.10:5060>
Call-ID: 33536115-beee-402b-acbe-0b29e11df6c0
CSeq: 44577 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.7.2
Content-Length: 0
<--- Received SIP response (1113 bytes) from UDP:176.12.16.218:46212 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.180.224.10:5060;rport=5060;received=207.180.224.10;branch=z9hG4bKPj5c49ef41-e7a5-4a72-bdd3-622302b828aa
Call-ID: 33536115-beee-402b-acbe-0b29e11df6c0
From: <sip:102@207.180.224.10>;tag=d30ee158-12c9-49f6-bc84-bf1793c7c29c
To: <sip:102@176.12.16.218;ob>;tag=z9hG4bKPj5c49ef41-e7a5-4a72-bdd3-622302b828aa
CSeq: 44577 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_ali_n-28/r2457
Content-Type: application/sdp
Content-Length: 280
v=0
o=- 3768817320 3768817320 IN IP4 10.0.0.162
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 99 0 8 101
c=IN IP4 10.0.0.162
a=sendrecv
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<--- Transmitting SIP request (431 bytes) to UDP:176.12.16.218:46212 --->
OPTIONS sip:102@176.12.16.218:46212;ob SIP/2.0
Via: SIP/2.0/UDP 207.180.224.10:5060;rport;branch=z9hG4bKPj5a2efbd1-fa56-4e00-ad1e-cf94ea901493
From: <sip:102@207.180.224.10>;tag=2a3f2292-45d7-4c1b-9f6d-ba4767d54a6b
To: <sip:102@176.12.16.218;ob>
Contact: <sip:102@207.180.224.10:5060>
Call-ID: b9e2aded-ca67-4763-bbab-fc38de61d690
CSeq: 34377 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.7.2
Content-Length: 0
<--- Received SIP response (1113 bytes) from UDP:176.12.16.218:46212 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.180.224.10:5060;rport=5060;received=207.180.224.10;branch=z9hG4bKPj5a2efbd1-fa56-4e00-ad1e-cf94ea901493
Call-ID: b9e2aded-ca67-4763-bbab-fc38de61d690
From: <sip:102@207.180.224.10>;tag=2a3f2292-45d7-4c1b-9f6d-ba4767d54a6b
To: <sip:102@176.12.16.218;ob>;tag=z9hG4bKPj5a2efbd1-fa56-4e00-ad1e-cf94ea901493
CSeq: 34377 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_ali_n-28/r2457
Content-Type: application/sdp
Content-Length: 280
v=0
o=- 3768817330 3768817330 IN IP4 10.0.0.162
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 99 0 8 101
c=IN IP4 10.0.0.162
a=sendrecv
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
vmi214442*CLI>
vmi214442*CLI>
vmi214442*CLI>
vmi214442*CLI> call here :
No such command 'call here :' (type 'core show help call here' for other possible commands)
vmi214442*CLI>
<--- Received SIP request (2081 bytes) from WSS:176.12.16.218:58448 --->
INVITE sip:1102@asterisk.org SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKmNLARwaHkbENLBFhbIXMuFJz0QUsKaBS;rport
From: "test"<sip:101@wrtc.bgtel.bg>;tag=yru9NokWbrRUUrrnLyhJ
To: <sip:1102@asterisk.org>
Contact: "test"<sips:101@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 962b8d80-fc62-e724-7597-381bb086d9db
CSeq: 29857 INVITE
Content-Type: application/sdp
Content-Length: 1512
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
v=0
o=mozilla...THIS_IS_SDPARTA-67.0 2418193502161601500 0 IN IP4 127.0.0.1
s=Doubango Telecom - firefox
t=0 0
a=sendrecv
a=fingerprint:sha-256 73:44:F6:F4:16:8F:8C:76:92:6B:89:65:EA:68:2F:B5:A0:E8:DB:3E:4E:3C:23:AF:8A:9F:97:45:47:C1:6B:C2
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 53693 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 176.12.16.218
a=candidate:0 1 UDP 2122252543 10.0.0.167 53693 typ host
a=candidate:5 1 TCP 2105524479 10.0.0.167 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 10.0.0.167 33475 typ host
a=candidate:5 2 TCP 2105524478 10.0.0.167 9 typ host tcptype active
a=candidate:2 1 UDP 1686052607 176.12.16.218 53693 typ srflx raddr 10.0.0.167 rport 53693
a=candidate:2 2 UDP 1686052606 176.12.16.218 33475 typ srflx raddr 10.0.0.167 rport 33475
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:17e0e0a85c3a7abe1f9920db1ac5b5bd
a=ice-ufrag:1cbf0051
a=mid:0
a=msid:{24e05874-eb5b-4697-92fe-2d1737216675} {54bc3c3a-a2b4-4b5e-ba97-0ac8ab1372db}
a=rtcp:33475 IN IP4 176.12.16.218
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:367690960 cname:{036d1371-c533-4abe-8fd2-8e3ac86a7b96}
<--- Transmitting SIP response (547 bytes) to WSS:176.12.16.218:58448 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=58448;received=176.12.16.218;branch=z9hG4bKmNLARwaHkbENLBFhbIXMuFJz0QUsKaBS
Call-ID: 962b8d80-fc62-e724-7597-381bb086d9db
From: "test" <sip:101@wrtc.bgtel.bg>;tag=yru9NokWbrRUUrrnLyhJ
To: <sip:1102@asterisk.org>;tag=z9hG4bKmNLARwaHkbENLBFhbIXMuFJz0QUsKaBS
CSeq: 29857 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1559828543/addc0a4a5b1eae668de588776982a920",opaque="597721ea67b01a43",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.7.2
Content-Length: 0
<--- Received SIP request (365 bytes) from WSS:176.12.16.218:58448 --->
ACK sip:1102@asterisk.org SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKmNLARwaHkbENLBFhbIXMuFJz0QUsKaBS;rport
From: "test"<sip:101@wrtc.bgtel.bg>;tag=yru9NokWbrRUUrrnLyhJ
To: <sip:1102@asterisk.org>;tag=z9hG4bKmNLARwaHkbENLBFhbIXMuFJz0QUsKaBS
Call-ID: 962b8d80-fc62-e724-7597-381bb086d9db
CSeq: 29857 ACK
Content-Length: 0
Max-Forwards: 70
<--- Received SIP request (2363 bytes) from WSS:176.12.16.218:58448 --->
INVITE sip:1102@asterisk.org SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKJXbhkfVAuEJIsNpA4ynQvylKp8WJ7Sis;rport
From: "test"<sip:101@wrtc.bgtel.bg>;tag=yru9NokWbrRUUrrnLyhJ
To: <sip:1102@asterisk.org>
Contact: "test"<sips:101@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 962b8d80-fc62-e724-7597-381bb086d9db
CSeq: 29858 INVITE
Content-Type: application/sdp
Content-Length: 1512
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="1559828543/addc0a4a5b1eae668de588776982a920",uri="sip:1102@asterisk.org",response="9e60385cd75464b25b19a42f7994f2d7",algorithm=md5,cnonce="81dc5088b8afb88785756d6c4f4512ee",opaque="597721ea67b01a43",qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
v=0
o=mozilla...THIS_IS_SDPARTA-67.0 2418193502161601500 0 IN IP4 127.0.0.1
s=Doubango Telecom - firefox
t=0 0
a=sendrecv
a=fingerprint:sha-256 73:44:F6:F4:16:8F:8C:76:92:6B:89:65:EA:68:2F:B5:A0:E8:DB:3E:4E:3C:23:AF:8A:9F:97:45:47:C1:6B:C2
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 53693 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 176.12.16.218
a=candidate:0 1 UDP 2122252543 10.0.0.167 53693 typ host
a=candidate:5 1 TCP 2105524479 10.0.0.167 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 10.0.0.167 33475 typ host
a=candidate:5 2 TCP 2105524478 10.0.0.167 9 typ host tcptype active
a=candidate:2 1 UDP 1686052607 176.12.16.218 53693 typ srflx raddr 10.0.0.167 rport 53693
a=candidate:2 2 UDP 1686052606 176.12.16.218 33475 typ srflx raddr 10.0.0.167 rport 33475
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:17e0e0a85c3a7abe1f9920db1ac5b5bd
a=ice-ufrag:1cbf0051
a=mid:0
a=msid:{24e05874-eb5b-4697-92fe-2d1737216675} {54bc3c3a-a2b4-4b5e-ba97-0ac8ab1372db}
a=rtcp:33475 IN IP4 176.12.16.218
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:367690960 cname:{036d1371-c533-4abe-8fd2-8e3ac86a7b96}
== Setting global variable 'SIPDOMAIN' to 'asterisk.org'
<--- Transmitting SIP response (351 bytes) to WSS:176.12.16.218:58448 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=58448;received=176.12.16.218;branch=z9hG4bKJXbhkfVAuEJIsNpA4ynQvylKp8WJ7Sis
Call-ID: 962b8d80-fc62-e724-7597-381bb086d9db
From: "test" <sip:101@wrtc.bgtel.bg>;tag=yru9NokWbrRUUrrnLyhJ
To: <sip:1102@asterisk.org>
CSeq: 29858 INVITE
Server: Asterisk PBX 15.7.2
Content-Length: 0
-- Executing [1102@default:1] Dial("PJSIP/101-00000001", "PJSIP/102") in new stack
[Jun 6 15:42:23] ERROR[14687]: chan_pjsip.c:2497 request: Failed to create outgoing session to endpoint '102'
[Jun 6 15:42:23] WARNING[14833][C-00000002]: app_dial.c:2512 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/101-00000001' status is 'CHANUNAVAIL'
== Contact 102/sip:102@176.12.16.218:46212;ob has been deleted
== Endpoint 102 is now Unreachable
<--- Transmitting SIP response (429 bytes) to WSS:176.12.16.218:58448 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=58448;received=176.12.16.218;branch=z9hG4bKJXbhkfVAuEJIsNpA4ynQvylKp8WJ7Sis
Call-ID: 962b8d80-fc62-e724-7597-381bb086d9db
From: "test" <sip:101@wrtc.bgtel.bg>;tag=yru9NokWbrRUUrrnLyhJ
To: <sip:1102@asterisk.org>;tag=76198bf1-3d0e-4285-af11-5492da017ee9
CSeq: 29858 INVITE
Server: Asterisk PBX 15.7.2
Reason: Q.850;cause=34
Content-Length: 0
<--- Received SIP request (362 bytes) from WSS:176.12.16.218:58448 --->
ACK sip:1102@asterisk.org SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKJXbhkfVAuEJIsNpA4ynQvylKp8WJ7Sis;rport
From: "test"<sip:101@wrtc.bgtel.bg>;tag=yru9NokWbrRUUrrnLyhJ
To: <sip:1102@asterisk.org>;tag=76198bf1-3d0e-4285-af11-5492da017ee9
Call-ID: 962b8d80-fc62-e724-7597-381bb086d9db
CSeq: 29858 ACK
Content-Length: 0
Max-Forwards: 70
vmi214442*CLI> pjsip show contacts
Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================
Contact: 101/sips:101@176.12.16.218:58448;transport=ws; eb17f306df Avail 71.086
Objects found: 1