Restored a backup, no phones will ring

So no phones will ring.

Call will go to voicemail if a did
if a queue they follow the rout of the queue

Phones are registered.

-- Executing [s@macro-user-callerid:49] Set("PJSIP/L3-use1a-0000000f", "CHANNEL(language)=en") in new stack
-- Executing [s@macro-vm:2] Set("PJSIP/L3-use1a-0000000f", "VMGAIN=") in new stack
-- Executing [s@macro-vm:3] Macro("PJSIP/L3-use1a-0000000f", "blkvm-check,") in new stack
-- Executing [s@macro-blkvm-check:1] Set("PJSIP/L3-use1a-0000000f", "GOSUB_RETVAL=") in new stack
-- Executing [s@macro-blkvm-check:2] ExecIf("PJSIP/L3-use1a-0000000f", "0?Set(GOSUB_RETVAL=TRUE)") in new stack
-- Executing [s@macro-blkvm-check:3] MacroExit("PJSIP/L3-use1a-0000000f", "") in new stack
-- Executing [s@macro-vm:4] GotoIf("PJSIP/L3-use1a-0000000f", "1?vmx,1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] Set("PJSIP/L3-use1a-0000000f", "MEXTEN=32897") in new stack
-- Executing [vmx@macro-vm:2] Set("PJSIP/L3-use1a-0000000f", "MMODE=NOANSWER") in new stack
-- Executing [vmx@macro-vm:3] Set("PJSIP/L3-use1a-0000000f", "RETVM=") in new stack
-- Executing [vmx@macro-vm:4] Set("PJSIP/L3-use1a-0000000f", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:5] Macro("PJSIP/L3-use1a-0000000f", "get-vmcontext,32897") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("PJSIP/L3-use1a-0000000f", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("PJSIP/L3-use1a-0000000f", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("PJSIP/L3-use1a-0000000f", "") in new stack
-- Executing [vmx@macro-vm:6] Set("PJSIP/L3-use1a-0000000f", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:7] NoOp("PJSIP/L3-use1a-0000000f", "MODE IS: unavail") in new stack
-- Executing [vmx@macro-vm:8] GotoIf("PJSIP/L3-use1a-0000000f", "1?chknomsg") in new stack
-- Goto (macro-vm,vmx,10)
-- Executing [vmx@macro-vm:10] GotoIf("PJSIP/L3-use1a-0000000f", "0?s-NOANSWER,1") in new stack
-- Executing [vmx@macro-vm:11] GotoIf("PJSIP/L3-use1a-0000000f", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,13)
-- Executing [vmx@macro-vm:13] NoOp("PJSIP/L3-use1a-0000000f", "Checking if ext 32897 is enabled: ") in new stack
-- Executing [vmx@macro-vm:14] GotoIf("PJSIP/L3-use1a-0000000f", "1?s-NOANSWER,1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-vm:1] Macro("PJSIP/L3-use1a-0000000f", "get-vmcontext,32897") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("PJSIP/L3-use1a-0000000f", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("PJSIP/L3-use1a-0000000f", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("PJSIP/L3-use1a-0000000f", "") in new stack
-- Executing [s-NOANSWER@macro-vm:2] VoiceMail("PJSIP/L3-use1a-0000000f", "32897@default,u") in new stack
-- <PJSIP/L3-use1a-0000000f> Playing '/opt/asterisk/tel40/spool/voicemail/default/32897/greet.gsm' (language 'en')

== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on ‘PJSIP/L3-use1a-0000000f’ in macro ‘vm’
== Spawn extension (macro-exten-vm, s, 23) exited non-zero on ‘PJSIP/L3-use1a-0000000f’ in macro ‘exten-vm’
== Spawn extension (ext-local, 32897, 3) exited non-zero on ‘PJSIP/L3-use1a-0000000f’
– Executing [h@ext-local:1] Macro(“PJSIP/L3-use1a-0000000f”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/L3-use1a-0000000f”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/L3-use1a-0000000f”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/L3-use1a-0000000f”, " montior file= ") in new stack
– Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/L3-use1a-0000000f”, “1?skipagi”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] Hangup(“PJSIP/L3-use1a-0000000f”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/L3-use1a-0000000f’

[root@ec2-tel40 asterisk]# asterisk -vx “pjsip show aor 32897”

  Aor:  <Aor..............................................>  <MaxContact>
Contact:  <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..>

==========================================================================================

  Aor:  32897                                                5
Contact:  32897/sip:[email protected]:5060           25ba9f6f0c Avail        67.983
Contact:  32897/sip:[email protected]:5061;rinstanc a13b567d49 Avail       133.710

ParameterName : ParameterValue

authenticate_qualify : false
contact : sip:[email protected]:5060
contact : sip:[email protected]:5061;rinstance=2ddb437706e127fc
default_expiration : 3600
mailboxes : 32897@default
max_contacts : 5
maximum_expiration : 3600
minimum_expiration : 60
outbound_proxy :
qualify_frequency : 60
qualify_timeout : 3.000000
remove_existing : false
support_path : false
voicemail_extension :

outbound calls work by the way.
DND is off,
Call waiting is enabled.

I see SDP message in console like

<— Received SIP response (399 bytes) from UDP:10.119.101.242:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.245.18:5060;rport=5060;branch=z9hG4bKPj97599e8f-8110-49c5-afef-c1e1c05e200a
Contact: sip:[email protected]:5061;transport=UDP
To: sip:[email protected];tag=4a248e3b
From: sip:[email protected];tag=72f7f3c9-c9f3-4e71-9b9f-ce62e2c4ef71
Call-ID: mMWeXBMqx5sh43scCwiQww…
CSeq: 9505 NOTIFY
User-Agent: Z 5.4.6 rv2.10.10.2
Content-Length: 0

<— Transmitting SIP request (489 bytes) to UDP:10.119.101.242:5061 —>
OPTIONS sip:[email protected]:5061;rinstance=2a0e2172292f0801 SIP/2.0
Via: SIP/2.0/UDP 10.123.245.18:5060;rport;branch=z9hG4bKPj0e14fa1f-a555-4baa-b614-26182a38d1be
From: sip:[email protected];tag=4b939a75-5c81-49b2-8551-9c608785f6ed
To: sip:[email protected];rinstance=2a0e2172292f0801
Contact: sip:[email protected]:5060
Call-ID: d451e2f9-f530-471b-9903-4df18105786e
CSeq: 9575 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.73(16.12.0)
Content-Length: 0

But I never see a big change or more messages when trying to call the exten.

I did capture

<— Transmitting SIP response (526 bytes) to UDP:10.119.101.242:5061 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.119.101.242:5061;rport=5061;received=10.119.101.242;branch=z9hG4bK-524287-1—4d95f8974e607599
Call-ID: JEwYF9omqrgE6duK4mDcrw…
From: sip:[email protected];tag=f14e1113
To: sip:[email protected];tag=z9hG4bK-524287-1—4d95f8974e607599
CSeq: 13 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1599602999/2a1c89c71ba116d01d000dd486df6390”,opaque=“41a321cb3d555456”,stale=true,algorithm=md5,qop=“auth”
Server: FPBX-15.0.16.73(16.12.0)
Content-Length: 0

<— Received SIP request (871 bytes) from UDP:10.119.101.242:5061 —>
REGISTER sip:10.123.245.18;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.119.101.242:5061;branch=z9hG4bK-524287-1—ca0205ae5432ff17
Max-Forwards: 70
Contact: sip:[email protected]:5061;rinstance=2a0e2172292f0801;transport=UDP
To: sip:[email protected];transport=UDP
From: sip:[email protected];transport=UDP;tag=f14e1113
Call-ID: JEwYF9omqrgE6duK4mDcrw…
CSeq: 14 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.4.6 rv2.10.10.2
Authorization: Digest username=“32897”,realm=“asterisk”,nonce=“1599602999/2a1c89c71ba116d01d000dd486df6390”,uri=“sip:10.123.245.18;transport=UDP”,response=“378bec666429919c1ae1129e9ed098f6”,cnonce=“0b223140034b360a3a044f987c1ca60d”,nc=00000001,qop=auth,algorithm=md5,opaque=“41a321cb3d555456”
Allow-Events: presence, kpml, talk
Content-Length: 0

<— Transmitting SIP response (542 bytes) to UDP:10.119.101.242:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.119.101.242:5061;rport=5061;received=10.119.101.242;branch=z9hG4bK-524287-1—ca0205ae5432ff17
Call-ID: JEwYF9omqrgE6duK4mDcrw…
From: sip:[email protected];tag=f14e1113
To: sip:[email protected];tag=z9hG4bK-524287-1—ca0205ae5432ff17
CSeq: 14 REGISTER
Date: Tue, 08 Sep 2020 22:09:59 GMT
Contact: sip:[email protected]:5060;expires=667
Contact: sip:[email protected]:5061;rinstance=2a0e2172292f0801;expires=59
Expires: 60
Server: FPBX-15.0.16.73(16.12.0)
Content-Length: 0

<— Received SIP request (908 bytes) from UDP:10.119.101.242:5061 —>
SUBSCRIBE sip:10.123.245.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.119.101.242:5061;branch=z9hG4bK-524287-1—8138477fe4ee6fb7
Max-Forwards: 70
Contact: sip:[email protected]:5061;transport=UDP
To: sip:[email protected];tag=72f7f3c9-c9f3-4e71-9b9f-ce62e2c4ef71
From: sip:[email protected];tag=4a248e3b
Call-ID: mMWeXBMqx5sh43scCwiQww…
CSeq: 8 SUBSCRIBE
Expires: 60
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.4.6 rv2.10.10.2
Authorization: Digest username=“32897”,realm=“asterisk”,nonce=“1599602679/5c00d1d3fcef20e9ea5ef03151333a99”,uri=“sip:10.123.245.18:5060”,response=“239a85efef288ec6cb62a954661c9ec6”,cnonce=“43db6700d56601bfb8c7bdeda38a3c60”,nc=00000007,qop=auth,algorithm=md5,opaque=“63f939c01924d2fe”
Event: message-summary
Allow-Events: presence, kpml, talk
Content-Length: 0

Its back to working ringing now. I don’t have audio on one direction on call incoming to the system. that is the next issue. There is no nat between nodes

Hi ! Go to Asterisk SIP Settings > See if all your local networks are there.

They are.

Whats odd is it seems like if you open the extension and than subitt it then it works.

Why would that be? I have close to 1K extension so not happy if I have to do that

Type this on your browser: http://your-pbx-ip:your-pbx-webgui-port/admin/config.php?display=extensions&action=resetall

That should act like going one by one and applying on them.

2 Likes

Thats a magical command.

Thanks.

We’ve done one backup/restore to get a v13 system to v15. We had at least two areas where the v15 GUI was showing the correct info, but the actual function was not correct until we did the same thing: Open the item, change something, change back, apply.

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