[asterisk-bugs] [JIRA] (ASTERISK-30028) Can't make outgoing calls : sip/2.0 401 unauthorized
Joshua C. Colp (JIRA)
noreply at issues.asterisk.org
Tue Apr 26 08:49:40 CDT 2022
[ https://issues.asterisk.org/jira/browse/ASTERISK-30028?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258905#comment-258905 ]
Joshua C. Colp commented on ASTERISK-30028:
-------------------------------------------
We appreciate the difficulties you are facing, however this does not appear to be a bug report and your request or comments would be better served in a different forum.
The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.
Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.
Thanks!
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
> Can't make outgoing calls : sip/2.0 401 unauthorized
> -----------------------------------------------------
>
> Key: ASTERISK-30028
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-30028
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Resources/res_pjsip
> Affects Versions: 16.25.2
> Environment: Ubuntu 20.04
> Reporter: Hery RARIVO
> Severity: Major
>
> Hi,
> I can't make outgoing calls.I thought it was my ISP but they say that there is no traffic from me. It's just ringing and after " Everyone is busy/congested at this time (1:0/0/1)
> So heres's my pjsip.conf :
> [transport-udp]
> type=transport
> protocol=udp
> bind=0.0.0.0
> local_net = 192.168.0.0/24
> external_media_address = 217.146.224.140
> external_signaling_address = 217.146.224.140
> [0174901008]
> type=endpoint
> context=sipmivoaka
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> aors=0174901008
> auth=0174901008
> [0174901008]
> type=aor
> max_contacts=1
> [0174901008]
> type=auth
> auth_type=userpass
> password=test
> username=0174901008
> Log :
> PJSIP Logging enabled
> <--- Received SIP request (1016 bytes) from UDP:192.168.0.22:22706 --->
> INVITE sip:0969363030 at 192.168.0.2 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-;rport
> Max-Forwards: 70
> Contact: <sip:0174901008 at 192.168.0.22:22706>
> To: "0969363030"<sip:0969363030 at 192.168.0.2>
> From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
> Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> Content-Type: application/sdp
> User-Agent: X-Lite release 1011s stamp 41150
> Content-Length: 467
> v=0
> o=- 5 2 IN IP4 192.168.0.22
> s=CounterPath X-Lite 3.0
> c=IN IP4 192.168.0.22
> t=0 0
> m=audio 3880 RTP/AVP 107 119 100 106 0 105 98 8 101
> a=alt:1 2 : Y7jmOD3P eVN2xzMi 192.168.0.22 3880
> a=alt:2 1 : ReUU43bs BHpj9E9S 192.168.56.1 3880
> a=fmtp:101 0-15
> a=rtpmap:107 BV32/16000
> a=rtpmap:119 BV32-FEC/16000
> a=rtpmap:100 SPEEX/16000
> a=rtpmap:106 SPEEX-FEC/16000
> a=rtpmap:105 SPEEX-FEC/8000
> a=rtpmap:98 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
> <--- Transmitting SIP response (559 bytes) to UDP:192.168.0.22:22706 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-
> Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
> From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
> To: "0969363030" <sip:0969363030 at 192.168.0.2>;tag=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-
> CSeq: 1 INVITE
> WWW-Authenticate: Digest realm="asterisk",nonce="1650980332/8708d9b872bc30e1f038e02c9451dce6",opaque="663078bc7460d60f",algorithm=md5,qop="auth"
> Server: Asterisk PBX 16.25.2
> Content-Length: 0
> <--- Received SIP request (365 bytes) from UDP:192.168.0.22:22706 --->
> ACK sip:0969363030 at 192.168.0.2 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-;rport
> To: "0969363030" <sip:0969363030 at 192.168.0.2>;tag=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-
> From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
> Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
> CSeq: 1 ACK
> Content-Length: 0
> <--- Received SIP request (1310 bytes) from UDP:192.168.0.22:22706 --->
> INVITE sip:0969363030 at 192.168.0.2 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-8220d018a045397b-1--d87543-;rport
> Max-Forwards: 70
> Contact: <sip:0174901008 at 192.168.0.22:22706>
> To: "0969363030"<sip:0969363030 at 192.168.0.2>
> From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
> Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
> CSeq: 2 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> Content-Type: application/sdp
> User-Agent: X-Lite release 1011s stamp 41150
> Authorization: Digest username="0174901008",realm="asterisk",nonce="1650980332/8708d9b872bc30e1f038e02c9451dce6",uri="sip:0969363030 at 192.168.0.2",response="573750b29d0db61028cd4b1c6fa44793",cnonce="557780aeb6d493f9e95a4e958c65383d",nc=00000001,qop=auth,algorithm=md5,opaque="663078bc7460d60f"
> Content-Length: 467
> v=0
> o=- 5 2 IN IP4 192.168.0.22
> s=CounterPath X-Lite 3.0
> c=IN IP4 192.168.0.22
> t=0 0
> m=audio 3880 RTP/AVP 107 119 100 106 0 105 98 8 101
> a=alt:1 2 : Y7jmOD3P eVN2xzMi 192.168.0.22 3880
> a=alt:2 1 : ReUU43bs BHpj9E9S 192.168.56.1 3880
> a=fmtp:101 0-15
> a=rtpmap:107 BV32/16000
> a=rtpmap:119 BV32-FEC/16000
> a=rtpmap:100 SPEEX/16000
> a=rtpmap:106 SPEEX-FEC/16000
> a=rtpmap:105 SPEEX-FEC/8000
> a=rtpmap:98 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
> <--- Transmitting SIP response (360 bytes) to UDP:192.168.0.22:22706 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-8220d018a045397b-1--d87543-
> Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
> From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
> To: "0969363030" <sip:0969363030 at 192.168.0.2>
> CSeq: 2 INVITE
> Server: Asterisk PBX 16.25.2
> Content-Length: 0
> -- Executing [0969363030 at sipmivoaka:1] Set("PJSIP/0174901008-00000000", "CALLERID(num)=0974901008") in new stack
> -- Executing [0969363030 at sipmivoaka:2] Gosub("PJSIP/0174901008-00000000", "my-gosub,s,1") in new stack
> -- Executing [s at my-gosub:1] NoOp("PJSIP/0174901008-00000000", "") in new stack
> -- Executing [s at my-gosub:2] Set("PJSIP/0174901008-00000000", "fname=/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s") in new stack
> -- Executing [s at my-gosub:3] Set("PJSIP/0174901008-00000000", "CDR(filename)=1650980332.0-2022-04-26-13_38-0974901008-s.mp3") in new stack
> -- Executing [s at my-gosub:4] Set("PJSIP/0174901008-00000000", "MONITOR_OPT=nice -n 19 /usr/bin/lame -b 16 -silent "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav" "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.mp3" && rm -f "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav"") in new stack
> -- Executing [s at my-gosub:5] MixMonitor("PJSIP/0174901008-00000000", "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav,b,nice -n 19 /usr/bin/lame -b 16 -silent "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav" "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.mp3" && rm -f "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav"") in new stack
> == Begin MixMonitor Recording PJSIP/0174901008-00000000
> -- Executing [s at my-gosub:6] Return("PJSIP/0174901008-00000000", "") in new stack
> -- Executing [0969363030 at sipmivoaka:3] Dial("PJSIP/0174901008-00000000", "PJSIP/0969363030 at axialys,40,tr") in new stack
> -- Called PJSIP/0969363030 at axialys
> <--- Transmitting SIP response (546 bytes) to UDP:192.168.0.22:22706 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-8220d018a045397b-1--d87543-
> Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
> From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
> To: "0969363030" <sip:0969363030 at 192.168.0.2>;tag=85aa64e6-f3c8-4101-b3da-84d092c117b7
> CSeq: 2 INVITE
> Server: Asterisk PBX 16.25.2
> Contact: <sip:192.168.0.2:5060>
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
> Content-Length: 0
> <--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
> INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
> Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
> From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
> To: <sip:0969363030 at sip-ng.axialys.net>
> Contact: <sip:asterisk at 217.146.224.140:5060>
> Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
> CSeq: 3600 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 16.25.2
> Content-Type: application/sdp
> Content-Length: 314
> v=0
> o=- 1086778031 1086778031 IN IP4 217.146.224.140
> s=Asterisk
> c=IN IP4 217.146.224.140
> t=0 0
> m=audio 18246 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
> INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
> Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
> From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
> To: <sip:0969363030 at sip-ng.axialys.net>
> Contact: <sip:asterisk at 217.146.224.140:5060>
> Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
> CSeq: 3600 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 16.25.2
> Content-Type: application/sdp
> Content-Length: 314
> v=0
> o=- 1086778031 1086778031 IN IP4 217.146.224.140
> s=Asterisk
> c=IN IP4 217.146.224.140
> t=0 0
> m=audio 18246 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
> INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
> Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
> From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
> To: <sip:0969363030 at sip-ng.axialys.net>
> Contact: <sip:asterisk at 217.146.224.140:5060>
> Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
> CSeq: 3600 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 16.25.2
> Content-Type: application/sdp
> Content-Length: 314
> v=0
> o=- 1086778031 1086778031 IN IP4 217.146.224.140
> s=Asterisk
> c=IN IP4 217.146.224.140
> t=0 0
> m=audio 18246 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
> INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
> Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
> From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
> To: <sip:0969363030 at sip-ng.axialys.net>
> Contact: <sip:asterisk at 217.146.224.140:5060>
> Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
> CSeq: 3600 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 16.25.2
> Content-Type: application/sdp
> Content-Length: 314
> v=0
> o=- 1086778031 1086778031 IN IP4 217.146.224.140
> s=Asterisk
> c=IN IP4 217.146.224.140
> t=0 0
> m=audio 18246 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
> INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
> Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
> From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
> To: <sip:0969363030 at sip-ng.axialys.net>
> Contact: <sip:asterisk at 217.146.224.140:5060>
> Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
> CSeq: 3600 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 16.25.2
> Content-Type: application/sdp
> Content-Length: 314
> v=0
> o=- 1086778031 1086778031 IN IP4 217.146.224.140
> s=Asterisk
> c=IN IP4 217.146.224.140
> t=0 0
> m=audio 18246 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
> INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
> Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
> From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
> To: <sip:0969363030 at sip-ng.axialys.net>
> Contact: <sip:asterisk at 217.146.224.140:5060>
> Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
> CSeq: 3600 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 16.25.2
> Content-Type: application/sdp
> Content-Length: 314
> v=0
> o=- 1086778031 1086778031 IN IP4 217.146.224.140
> s=Asterisk
> c=IN IP4 217.146.224.140
> t=0 0
> m=audio 18246 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <--- Received SIP request (548 bytes) from UDP:192.168.0.22:22706 --->
> SUBSCRIBE sip:0174901008 at 192.168.0.2 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-462e1405613ff014-1--d87543-;rport
> Max-Forwards: 70
> Contact: <sip:0174901008 at 192.168.0.22:22706>
> To: <sip:0174901008 at 192.168.0.2>
> From: <sip:0174901008 at 192.168.0.2>;tag=3b387b37
> Call-ID: ZTU4ZDY3ZjQxOTEwMzM4NWQ1N2QzYWM4ZjI5ZTFiYzA.
> CSeq: 1 SUBSCRIBE
> Expires: 300
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> User-Agent: X-Lite release 1011s stamp 41150
> Event: message-summary
> Content-Length: 0
> <--- Transmitting SIP response (549 bytes) to UDP:192.168.0.22:22706 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-462e1405613ff014-1--d87543-
> Call-ID: ZTU4ZDY3ZjQxOTEwMzM4NWQ1N2QzYWM4ZjI5ZTFiYzA.
> From: <sip:0174901008 at 192.168.0.2>;tag=3b387b37
> To: <sip:0174901008 at 192.168.0.2>;tag=z9hG4bK-d87543-462e1405613ff014-1--d87543-
> CSeq: 1 SUBSCRIBE
> WWW-Authenticate: Digest realm="asterisk",nonce="1650980354/4cecd3f7fbca64f7f5411d7f6ec4d7d7",opaque="4be95f0921e61bb6",algorithm=md5,qop="auth"
> Server: Asterisk PBX 16.25.2
> Content-Length: 0
> <--- Received SIP request (842 bytes) from UDP:192.168.0.22:22706 --->
> SUBSCRIBE sip:0174901008 at 192.168.0.2 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-332e8d3ff6149171-1--d87543-;rport
> Max-Forwards: 70
> Contact: <sip:0174901008 at 192.168.0.22:22706>
> To: <sip:0174901008 at 192.168.0.2>
> From: <sip:0174901008 at 192.168.0.2>;tag=3b387b37
> Call-ID: ZTU4ZDY3ZjQxOTEwMzM4NWQ1N2QzYWM4ZjI5ZTFiYzA.
> CSeq: 2 SUBSCRIBE
> Expires: 300
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> User-Agent: X-Lite release 1011s stamp 41150
> Authorization: Digest username="0174901008",realm="asterisk",nonce="1650980354/4cecd3f7fbca64f7f5411d7f6ec4d7d7",uri="sip:0174901008 at 192.168.0.2",response="06da050976be81aa74a3d06c55dd6f5a",cnonce="f89f148ffcb3974cbe9e0e3fab47acdf",nc=00000001,qop=auth,algorithm=md5,opaque="4be95f0921e61bb6"
> Event: message-summary
> Content-Length: 0
> <--- Transmitting SIP response (400 bytes) to UDP:192.168.0.22:22706 --->
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-332e8d3ff6149171-1--d87543-
> Call-ID: ZTU4ZDY3ZjQxOTEwMzM4NWQ1N2QzYWM4ZjI5ZTFiYzA.
> From: <sip:0174901008 at 192.168.0.2>;tag=3b387b37
> To: <sip:0174901008 at 192.168.0.2>;tag=z9hG4bK-d87543-332e8d3ff6149171-1--d87543-
> CSeq: 2 SUBSCRIBE
> Server: Asterisk PBX 16.25.2
> Content-Length: 0
> <--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
> INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
> Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
> From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
> To: <sip:0969363030 at sip-ng.axialys.net>
> Contact: <sip:asterisk at 217.146.224.140:5060>
> Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
> CSeq: 3600 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 16.25.2
> Content-Type: application/sdp
> Content-Length: 314
> v=0
> o=- 1086778031 1086778031 IN IP4 217.146.224.140
> s=Asterisk
> c=IN IP4 217.146.224.140
> t=0 0
> m=audio 18246 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> == Everyone is busy/congested at this time (1:0/0/1)
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