503 Service Unavailable


(Ruslan Griban) #1

After updating the modules yesterday, Freepbh stopped making calls

Цитата
2020/04/20 12:53:19.474513 192.168.15.12:5060 -> 192.168.75.243:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-d4ca573732c93684b3c026060e24d4f7;received=192.168.75.243
From: “999” sip:999@192.168.15.12;tag=7062d6bf
To: sip:708@192.168.15.12;tag=as7120ead7
Call-ID: bf9eccc9328bbc4bb3b3c81ff9bb2e63@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: FPBX-14.0.13.28(14.7.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Цитата
freepbx*CLI> sip set debug peer 999
SIP Debugging Enabled for IP: 192.168.75.243
– Registered SIP ‘783’ at 192.168.13.183:1061
> Saved useragent “Grandstream GXP1620 1.0.4.55” for peer 783
– Registered SIP ‘781’ at 192.168.13.185:1061
> Saved useragent “Grandstream GXP1620 1.0.4.55” for peer 781

<— SIP read from UDP:192.168.75.243:5060 —>
OPTIONS sip:192.168.15.12 SIP/2.0
Call-ID: 08f3fb969d0d573fd8069608711d9b54@0:0:0:0:0:0:0:0
CSeq: 81 OPTIONS
From: “999” sip:999@192.168.15.12;tag=9359419b
To: “999” sip:999@192.168.15.12
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-1f584bc4503b930806a203e880dd3cf9
Max-Forwards: 70
Contact: “999” sip:999@192.168.75.243:5060;transport=udp;registering_acc=192_168_15_12
User-Agent: Jitsi2.10.5550Windows 10
Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.75.243:5060 (NAT)
Looking for s in from-sip-external (domain 192.168.15.12)

<— Transmitting (NAT) to 192.168.75.243:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-1f584bc4503b930806a203e880dd3cf9;received=192.168.75.243;rport=5060
From: “999” sip:999@192.168.15.12;tag=9359419b
To: “999” sip:999@192.168.15.12;tag=as546cf425
Call-ID: 08f3fb969d0d573fd8069608711d9b54@0:0:0:0:0:0:0:0
CSeq: 81 OPTIONS
Server: FPBX-14.0.13.28(14.7.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:192.168.15.12:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘08f3fb969d0d573fd8069608711d9b54@0:0:0:0:0:0:0:0’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘1f2ce7d12e385b61e08e4d965568e7e0@0:0:0:0:0:0:0:0’ Method: OPTIONS
– Remote UNIX connection
– Remote UNIX connection disconnected

<— SIP read from UDP:192.168.75.243:5060 —>
INVITE sip:708@192.168.15.12 SIP/2.0
Call-ID: ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: “999” sip:999@192.168.15.12;tag=ab235d6c
To: sip:708@192.168.15.12
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-ce198a11a3c4f0e7693f80d3cc84e7be
Max-Forwards: 70
Contact: “999” sip:999@192.168.75.243:5060;transport=udp;registering_acc=192_168_15_12
User-Agent: Jitsi2.10.5550Windows 10
Content-Type: application/sdp
Content-Length: 909

v=0
o=999-jitsi.org 0 0 IN IP4 192.168.75.243
s=-
c=IN IP4 192.168.75.243
t=0 0
m=audio 5057 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=ptime:20
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5059 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[1:1920],y=[1:1080]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[1:1920],y=[1:1080]]
a=rtcp:5061
<------------->
— (11 headers 32 lines) —
Sending to 192.168.75.243:5060 (NAT)
Sending to 192.168.75.243:5060 (NAT)
Using INVITE request as basis request - ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
Found peer ‘999’ for ‘999’ from 192.168.75.243:5060

<— Reliably Transmitting (no NAT) to 192.168.75.243:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-ce198a11a3c4f0e7693f80d3cc84e7be;received=192.168.75.243
From: “999” sip:999@192.168.15.12;tag=ab235d6c
To: sip:708@192.168.15.12;tag=as385ea5a2
Call-ID: ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
Server: FPBX-14.0.13.28(14.7.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3579e35a”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.75.243:5060 —>
ACK sip:708@192.168.15.12 SIP/2.0
Call-ID: ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: “999” sip:999@192.168.15.12;tag=ab235d6c
To: sip:708@192.168.15.12;tag=as385ea5a2
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-ce198a11a3c4f0e7693f80d3cc84e7be
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.75.243:5060 —>
INVITE sip:708@192.168.15.12 SIP/2.0
Call-ID: ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: “999” sip:999@192.168.15.12;tag=ab235d6c
To: sip:708@192.168.15.12
Max-Forwards: 70
Contact: “999” sip:999@192.168.75.243:5060;transport=udp;registering_acc=192_168_15_12
User-Agent: Jitsi2.10.5550Windows 10
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-8f3960c698b279d965a1aec484abc3c0
Authorization: Digest username=“999”,realm=“asterisk”,nonce=“3579e35a”,uri="sip:708@192.168.15.12",response=“e88adde379d1055ee9a3747c5c0eda15”,algorithm=MD5
Content-Length: 909

v=0
o=999-jitsi.org 0 0 IN IP4 192.168.75.243
s=-
c=IN IP4 192.168.75.243
t=0 0
m=audio 5057 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=ptime:20
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5059 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[1:1920],y=[1:1080]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[1:1920],y=[1:1080]]
a=rtcp:5061
<------------->
— (12 headers 32 lines) —
Sending to 192.168.75.243:5060 (no NAT)
Using INVITE request as basis request - ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
Found peer ‘999’ for ‘999’ from 192.168.75.243:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 9
Found RTP audio format 100
Found RTP audio format 102
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 103
Found RTP audio format 3
Found RTP audio format 104
Found RTP audio format 101
Found audio description format opus for ID 96
Found audio description format SILK for ID 97
Found audio description format SILK for ID 98
Found audio description format G722 for ID 9
Found audio description format speex for ID 100
Found audio description format speex for ID 102
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 103
Found audio description format GSM for ID 3
Found audio description format speex for ID 104
Found audio description format telephone-event for ID 101
Found RTP video format 105
Found RTP video format 99
Found video description format H264 for ID 105
Found video description format H264 for ID 99
Capabilities: us - (ulaw|alaw|gsm|g726|g722|g723|g729), peer - audio=(ulaw|gsm|alaw|g722|opus|silk24|silk16|speex32|speex16|ilbc|speex)/video=(h264|red)/text=(nothing), combined - (ulaw|alaw|gsm|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f39a404a740 – Strict RTP learning after remote address set to: 192.168.75.243:5057
Peer audio RTP is at port 192.168.75.243:5057
Looking for 708 in from-internal (domain 192.168.15.12)
sip_route_dump: route/path hop: sip:999@192.168.75.243:5060;transport=udp;registering_acc=192_168_15_12

<— Transmitting (no NAT) to 192.168.75.243:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-8f3960c698b279d965a1aec484abc3c0;received=192.168.75.243
From: “999” sip:999@192.168.15.12;tag=ab235d6c
To: sip:708@192.168.15.12
Call-ID: ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: FPBX-14.0.13.28(14.7.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:708@192.168.15.12:5060
Content-Length: 0

<------------>
– Executing [708@from-internal:1] Macro(“SIP/999-00000009”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/999-00000009”, “TOUCH_MONITOR=1587377042.9”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/999-00000009”, “AMPUSER=999”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/999-00000009”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/999-00000009”, “1?Set(REALCALLERIDNUM=999)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/999-00000009”, “AMPUSER=999”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/999-00000009”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/999-00000009”, “AMPUSERCIDNAME=Admin”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“SIP/999-00000009”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“SIP/999-00000009”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/999-00000009”, “AMPUSERCID=999”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/999-00000009”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“SIP/999-00000009”, “CALLERID(all)=“Admin” <999>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“SIP/999-00000009”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/999-00000009”, “1?Set(GROUP(concurrency_limit)=999)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“SIP/999-00000009”, “1?Set(CHANNEL(language)=ru)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“SIP/999-00000009”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“SIP/999-00000009”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] GotoIf(“SIP/999-00000009”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“SIP/999-00000009”, “CALLERID(number)=999”) in new stack
– Executing [s@macro-user-callerid:38] Set(“SIP/999-00000009”, “CALLERID(name)=Admin”) in new stack
– Executing [s@macro-user-callerid:39] Set(“SIP/999-00000009”, “CALLERID(name)=Admin”) in new stack
– Executing [s@macro-user-callerid:40] Set(“SIP/999-00000009”, “CALLERID(name)=Admin”) in new stack
– Executing [s@macro-user-callerid:41] Set(“SIP/999-00000009”, “CALLERID(name)=Admin”) in new stack
– Executing [s@macro-user-callerid:42] ExecIf(“SIP/999-00000009”, “0?Set(CALLERID(number)=Admin)”) in new stack
– Executing [s@macro-user-callerid:43] Set(“SIP/999-00000009”, “CALLERID(number)=999”) in new stack
– Executing [s@macro-user-callerid:44] GotoIf(“SIP/999-00000009”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:45] Set(“SIP/999-00000009”, “CDR(cnam)=Admin”) in new stack
– Executing [s@macro-user-callerid:46] Set(“SIP/999-00000009”, “CDR(cnum)=999”) in new stack
– Executing [s@macro-user-callerid:47] Set(“SIP/999-00000009”, “CHANNEL(language)=ru”) in new stack
– Executing [s@macro-user-callerid:48] Congestion(“SIP/999-00000009”, “20”) in new stack

<— Reliably Transmitting (no NAT) to 192.168.75.243:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-8f3960c698b279d965a1aec484abc3c0;received=192.168.75.243
From: “999” sip:999@192.168.15.12;tag=ab235d6c
To: sip:708@192.168.15.12;tag=as69fc9558
Call-ID: ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: FPBX-14.0.13.28(14.7.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2020-04-20 13:04:02] WARNING[22412][C-0000000a]: channel.c:5005 ast_prod: Prodding channel ‘SIP/999-00000009’ failed
== Spawn extension (macro-user-callerid, s, 48) exited non-zero on ‘SIP/999-00000009’ in macro ‘user-callerid’
== Spawn extension (from-internal, 708, 1) exited non-zero on ‘SIP/999-00000009’
– Executing [h@from-internal:1] Macro(“SIP/999-00000009”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/999-00000009”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/999-00000009”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] System(“SIP/999-00000009”, “/var/lib/asterisk/bin/check_missed_call_answ ‘’ &”) in new stack

<— SIP read from UDP:192.168.75.243:5060 —>
ACK sip:708@192.168.15.12 SIP/2.0
Call-ID: ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: “999” sip:999@192.168.15.12;tag=ab235d6c
To: sip:708@192.168.15.12;tag=as69fc9558
Via: SIP/2.0/UDP 192.168.75.243:5060;branch=z9hG4bK-363239-8f3960c698b279d965a1aec484abc3c0
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
– Executing [s@macro-hangupcall:5] NoOp(“SIP/999-00000009”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:6] AGI(“SIP/999-00000009”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <SIP/999-00000009>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:7] Hangup(“SIP/999-00000009”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/999-00000009’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/999-00000009’
Really destroying SIP dialog ‘ed93da88ffe86dcd388137dbb0e0331b@0:0:0:0:0:0:0:0’ Method: ACK
[2020-04-20 13:04:03] NOTICE[16312]: chan_sip.c:17336 check_auth: Correct auth, but based on stale nonce received from ‘“0930463303” sip:0930463303@192.168.15.12;tag=691961361’
[2020-04-20 13:04:03] NOTICE[16312]: chan_sip.c:17336 check_auth: Correct auth, but based on stale nonce received from ‘“0952909103” sip:0952909103@192.168.15.12;tag=449713087’
[2020-04-20 13:04:03] NOTICE[16312]: chan_sip.c:17336 check_auth: Correct auth, but based on stale nonce received from ‘“0950103103” sip:0950103103@192.168.15.12;tag=167139800’
Really destroying SIP dialog ‘7d0860f34f6cd3c8903c62a5b98a3324@0:0:0:0:0:0:0:0’ Method: OPTIONS


(Ruslan Griban) #2

Solution: downgrade the Base module version to (14.0.28.35)


(system) closed #3

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