| Summary: | ASTERISK-30028: Can't make outgoing calls : sip/2.0 401 unauthorized | ||
| Reporter: | Hery RARIVO (hery2022) | Labels: | |
| Date Opened: | 2022-04-26 08:43:36 | Date Closed: | 2022-04-26 08:48:19 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Resources/res_pjsip |
| Versions: | 16.25.2 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Ubuntu 20.04 | Attachments: | |
| Description: | 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@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@192.168.0.22:22706> To: "0969363030"<sip:0969363030@192.168.0.2> From: <sip:0174901008@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@192.168.0.2>;tag=a463f547 To: "0969363030" <sip:0969363030@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@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@192.168.0.2>;tag=z9hG4bK-d87543-291fa2309f5d423c-1--d87543- From: <sip:0174901008@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@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@192.168.0.22:22706> To: "0969363030"<sip:0969363030@192.168.0.2> From: <sip:0174901008@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@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@192.168.0.2>;tag=a463f547 To: "0969363030" <sip:0969363030@192.168.0.2> CSeq: 2 INVITE Server: Asterisk PBX 16.25.2 Content-Length: 0 -- Executing [0969363030@sipmivoaka:1] Set("PJSIP/0174901008-00000000", "CALLERID(num)=0974901008") in new stack -- Executing [0969363030@sipmivoaka:2] Gosub("PJSIP/0174901008-00000000", "my-gosub,s,1") in new stack -- Executing [s@my-gosub:1] NoOp("PJSIP/0174901008-00000000", "") in new stack -- Executing [s@my-gosub:2] Set("PJSIP/0174901008-00000000", "fname=/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s") in new stack -- Executing [s@my-gosub:3] Set("PJSIP/0174901008-00000000", "CDR(filename)=1650980332.0-2022-04-26-13_38-0974901008-s.mp3") in new stack -- Executing [s@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@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@my-gosub:6] Return("PJSIP/0174901008-00000000", "") in new stack -- Executing [0969363030@sipmivoaka:3] Dial("PJSIP/0174901008-00000000", "PJSIP/0969363030@axialys,40,tr") in new stack -- Called PJSIP/0969363030@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@192.168.0.2>;tag=a463f547 To: "0969363030" <sip:0969363030@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@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@sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589 To: <sip:0969363030@sip-ng.axialys.net> Contact: <sip:asterisk@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@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@sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589 To: <sip:0969363030@sip-ng.axialys.net> Contact: <sip:asterisk@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@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@sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589 To: <sip:0969363030@sip-ng.axialys.net> Contact: <sip:asterisk@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@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@sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589 To: <sip:0969363030@sip-ng.axialys.net> Contact: <sip:asterisk@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@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@sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589 To: <sip:0969363030@sip-ng.axialys.net> Contact: <sip:asterisk@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@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@sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589 To: <sip:0969363030@sip-ng.axialys.net> Contact: <sip:asterisk@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@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@192.168.0.22:22706> To: <sip:0174901008@192.168.0.2> From: <sip:0174901008@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@192.168.0.2>;tag=3b387b37 To: <sip:0174901008@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@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@192.168.0.22:22706> To: <sip:0174901008@192.168.0.2> From: <sip:0174901008@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@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@192.168.0.2>;tag=3b387b37 To: <sip:0174901008@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@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@sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589 To: <sip:0969363030@sip-ng.axialys.net> Contact: <sip:asterisk@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) | ||
| Comments: | By: Asterisk Team (asteriskteam) 2022-04-26 08:43:37.125-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Joshua C. Colp (jcolp) 2022-04-26 08:48:08.551-0500 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 | ||