Keep getting sent to vmail

General Help
1

All my calls inbound go straight to vmail and we cannot make outbound calls
I turned on sip debug and got this

Anyone know how to get this working so I can make and take calls?

pbx*CLI> sip debug
SIP Debugging enabled
The ‘sip debug’ command is deprecated and will be removed in a future release. Please use ‘sip set debug’ instead.
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.2.142.30:5060:
REGISTER sip:inbound4.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 69.64.59.250:5060;branch=z9hG4bK6ac753f7;rport
From: sip:[email protected];tag=as40905d73
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 122 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“user”, realm=“asterisk”, algorithm=MD5, uri=“sip:inbound4.vitelity.net”, nonce=“1a09622f”, response="cc1ed3fdc2db0316c02343743e3e480e"
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0

pbx*CLI>
<— SIP read from 64.2.142.30:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 69.64.59.250:5060;branch=z9hG4bK6ac753f7;received=69.64.59.250;rport=5060
From: sip:[email protected];tag=as40905d73
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 122 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[email protected]
Content-Length: 0

<------------->
— (10 headers 0 lines) —
pbx*CLI>
<— SIP read from 64.2.142.30:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 69.64.59.250:5060;branch=z9hG4bK6ac753f7;received=69.64.59.250;rport=5060
From: sip:[email protected];tag=as40905d73
To: sip:[email protected];tag=as3f1cb1d3
Call-ID: [email protected]
CSeq: 122 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1a4486ec"
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Responding to challenge, registration to domain/host name inbound4.vitelity.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.2.142.30:5060:
REGISTER sip:inbound4.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 69.64.59.250:5060;branch=z9hG4bK2c22db99;rport
From: sip:[email protected];tag=as42e71bbb
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 123 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“user”, realm=“asterisk”, algorithm=MD5, uri=“sip:inbound4.vitelity.net”, nonce=“1a4486ec”, response="546b460e85f10876660f1179f935d51f"
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0

pbx*CLI>
<— SIP read from 64.2.142.30:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 69.64.59.250:5060;branch=z9hG4bK2c22db99;received=69.64.59.250;rport=5060
From: sip:[email protected];tag=as42e71bbb
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 123 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[email protected]
Content-Length: 0

<------------->
— (10 headers 0 lines) —
pbx*CLI>
<— SIP read from 64.2.142.30:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.64.59.250:5060;branch=z9hG4bK2c22db99;received=69.64.59.250;rport=5060
From: sip:[email protected];tag=as42e71bbb
To: sip:[email protected];tag=as3f1cb1d3
Call-ID: [email protected]
CSeq: 123 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: sip:[email protected];expires=60
Date: Wed, 07 Jan 2009 18:45:00 GMT
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

<— SIP read from 64.2.142.30:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK28597977;rport
From: “GSA/FTS/6TTP AT” sip:[email protected];tag=as10e7067f
To: sip:[email protected]
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 07 Jan 2009 18:45:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 4115 4115 IN IP4 64.2.142.30
s=session
c=IN IP4 64.2.142.30
t=0 0
m=audio 19956 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
— (13 headers 14 lines) —
Sending to 64.2.142.30 : 5060 (NAT)
Using INVITE request as basis request - [email protected]
Found no matching peer or user for '64.2.142.30:5060’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 64.2.142.30:19956
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 64.2.142.30:19956
Looking for 8167332028 in from-sip-external (domain 69.64.59.250)
list_route: hop: sip:[email protected]

<— Transmitting (NAT) to 64.2.142.30:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK28597977;received=64.2.142.30;rport=5060
From: “GSA/FTS/6TTP AT” sip:[email protected];tag=as10e7067f
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
– Executing [8167332028@from-sip-external:1] NoOp(“SIP/64.2.142.30-0884f098”, “Received incoming SIP connection from unknown peer to 8167332028”) in new stack
– Executing [8167332028@from-sip-external:2] Set(“SIP/64.2.142.30-0884f098”, “DID=8167332028”) in new stack
– Executing [8167332028@from-sip-external:3] Goto(“SIP/64.2.142.30-0884f098”, “s|1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/64.2.142.30-0884f098”, “1?from-trunk|8167332028|1”) in new stack
– Goto (from-trunk,8167332028,1)
– Executing [8167332028@from-trunk:1] Set(“SIP/64.2.142.30-0884f098”, “__FROM_DID=8167332028”) in new stack
– Executing [8167332028@from-trunk:2] Gosub(“SIP/64.2.142.30-0884f098”, “app-blacklist-check|s|1”) in new stack
– Executing [s@app-blacklist-check:1] LookupBlacklist(“SIP/64.2.142.30-0884f098”, “”) in new stack
– Executing [s@app-blacklist-check:2] GotoIf(“SIP/64.2.142.30-0884f098”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/64.2.142.30-0884f098”, “”) in new stack
– Executing [8167332028@from-trunk:3] Gosub(“SIP/64.2.142.30-0884f098”, “cidlookup|cidlookup_1|1”) in new stack
– Executing [cidlookup_1@cidlookup:1] Set(“SIP/64.2.142.30-0884f098”, “CALLERID(name)=Kansas City, MO Land Line”) in new stack
– Executing [cidlookup_1@cidlookup:2] Return(“SIP/64.2.142.30-0884f098”, “”) in new stack
– Executing [8167332028@from-trunk:4] ExecIf(“SIP/64.2.142.30-0884f098”, “0 |Set|CALLERID(name)=8168232055”) in new stack
– Executing [8167332028@from-trunk:5] SetMusicOnHold(“SIP/64.2.142.30-0884f098”, “none”) in new stack
– Executing [8167332028@from-trunk:6] Set(“SIP/64.2.142.30-0884f098”, “__MOHCLASS=none”) in new stack
– Executing [8167332028@from-trunk:7] Set(“SIP/64.2.142.30-0884f098”, “FAX_RX=disabled”) in new stack
– Executing [8167332028@from-trunk:8] Set(“SIP/64.2.142.30-0884f098”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [8167332028@from-trunk:9] SetCallerPres(“SIP/64.2.142.30-0884f098”, “allowed_not_screened”) in new stack
– Executing [8167332028@from-trunk:10] Goto(“SIP/64.2.142.30-0884f098”, “from-did-direct|501|1”) in new stack
– Goto (from-did-direct,501,1)
– Executing [501@from-did-direct:1] Set(“SIP/64.2.142.30-0884f098”, “__RINGTIMER=23”) in new stack
– Executing [501@from-did-direct:2] Macro(“SIP/64.2.142.30-0884f098”, “exten-vm|501|501”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“e[1 ;35;40mSIP/64.2.142.30-0884f098”, “user-callerid”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/64.2.142.30-0884f098”, “AMPUSER=8168232055”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/64.2.142.30-0884f098”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/64.2.142.30-0884f098”, “1|Set|REALCALLERIDNUM=8168232055”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/64.2.142.30-0884f098”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/64.2.142.30-0884f098”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“e[1;35 ;40mSIP/64.2.142.30-0884f098”, “1?report”) in new stack
– Goto (macro-user-callerid,s,11)
– Executing [s@macro-user-callerid:11] GotoIf(“SIP/64.2.142.30-0884f098”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:12] Set(“SIP/64.2.142.30-0884f098”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“SIP/64.2.142.30-0884f098”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,20)
– Executing [s@macro-user-callerid:20] NoOp(“SIP/64.2.142.30-0884f098”, “Using CallerID “Kansas City, MO Land Line” <8168232055>”) in new stack
– Executing [s@macro-exten-vm:2] Set(“SIP/64.2.142.30-0884f098”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“SIP/64.2.142.30-0884f098”, “VMBOX=501”) in new stack
– Executing [s@macro-exten-vm:4] Set(“SIP/64.2.142.30-0884f098”, “EXTTOCALL=501”) in new stack
– Executing [s@macro-exten-vm:5] Set(“SIP/64.2.142.30-0884f098”, “CFUEXT=”) in new stack
– Executing [s@macro-exten-vm:6] Set(“SIP/64.2.142.30-0884f098”, “CFBEXT=”) in new stack
– Executing [s@macro-exten-vm:7] Set(“SIP/64.2.142.30-0884f098”, “RT=23”) in new stack
– Executing [s@macro-exten-vm:8] Macro(“SIP/64.2.142.30-0884f098”, “record-enable|501|IN”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/64.2.142.30-0884f098”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/64.2.142.30-0884f098”, “recordingcheck|20090107-124518|1231353914.26”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090107-124518|1231353914.26: Inbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] MacroExit(“SIP/64.2.142.30-0884f098”, “”) in new stack
– Executing [s@macro-exten-vm:9] Macro(“SIP/64.2.142.30-0884f098”, “dial|23|trW|501”) in new stack
– Executing [s@macro-dial:1] GotoIf(“SIP/64.2.142.30-0884f098”, “0?diale[0;37;40 m”) in new stack
– Executing [s@macro-dial:2] SetMusicOnHold(“SIP/64.2.142.30-0884f098”, “none”) in new stack
– Executing [s@macro-dial:3] AGI(“SIP/64.2.142.30-0884f098”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
dialparties.agi: Caller ID name is ‘Kansas City, MO Land Line’ number is '8168232055’
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 501 to extension map
– dialparties.agi: Extension 501 cf is disabled
– dialparties.agi: Extension 501 do not disturb is disabled
– dialparties.agi: dbset CALLTRACE/501 to 8168232055
– dialparties.agi: Filtered ARG3: 501
– AGI Script dialparties.agi completed, returning 0
– Executing [s@macro-dial:7] Dial(“SIP/64.2.142.30-0884f098”, “SIP/501|23|trW”) in new stack
== Manager ‘admin’ logged off from 127.0.0.1
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dial:8] Set(“SIP/64.2.142.30-0884f098”, “DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-dial:9] GosubIf(“SIP/64.2.142.30-0884f098”, “0?CHANUNAVAIL|1”) in new stack
– Executing [s@macro-exten-vm:10] GotoIf(“SIP/64.2.142.30-0884f098”, “0?exit|return”) in new stack
– Executing [s@macro-exten-vm:11] Set(“SIP/64.2.142.30-0884f098”, “SV_DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:12] GosubIf(“SIP/64.2.142.30-0884f098”, “0?docfu|1”) in new stack
– Executing [s@macro-exten-vm:13] e[1; 36;40mGosubIf(“SIP/64.2.142.30-0884f098”, “0?docfb|1”) in new stack
– Executing [s@macro-exten-vm:14] Set(“SIP/64.2.142.30-0884f098”, “DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:15] NoOp(“SIP/64.2.142.30-0884f098”, “Voicemail is 501”) in new stack
– Executing [s@macro-exten-vm:16] GotoIf(“SIP/64.2.142.30-0884f098”, “0?s-CHANUNAVAIL|1”) in new stack
– Executing [s@macro-exten-vm:17] NoOp(“SIP/64.2.142.30-0884f098”, “Sending to Voicemail box 501”) in new stack
– Executing [s@macro-exten-vm:18] Macro(“SIP/64.2.142.30-0884f098”, “vm|501|CHANUNAVAIL|”) in new stack
– Executing [s@macro-vm:1] Macroe[0; 37;40m(“SIP/64.2.142.30-0884f098”, “user-callerid|SKIPTTL”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/64.2.142.30-0884f098”, “AMPUSER=8168232055”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/64.2.142.30-0884f098”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIfe[0; 37;40m(“SIP/64.2.142.30-0884f098”, “0|Set|REALCALLERIDNUM=8168232055”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/64.2.142.30-0884f098”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/64.2.142.30-0884f098”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/64.2.142.30-0884f098”, “1?report”) in new stack
– Goto (macro-user-callerid,s,11)
– Executing [s@macro-user-callerid:11] GotoIf(“SIP/64.2.142.30-0884f098”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,20)
– Executing [s@macro-user-callerid:20] NoOp(“SIP/64.2.142.30-0884f098”, “Using CallerID “Kansas City, MO Land Line” <8168232055>”) in new stack
– Executing [s@macro-vm:2] Set(“SIP/64.2.142.30-0884f098”, “VMGAIN=”"") in new stack
– Executing [s@macro-vm:3] GotoIf(“SIP/64.2.142.30-0884f098”, “1?vmx|1”) in new stack
– Goto (macro-vm,vmx,1)
– Executing [vmx@macro-vm:1] GotoIf(“SIP/64.2.142.30-0884f098”, “0?s-CHANUNAVAIL|1”) in new stack
– Executing [vmx@macro-vm:2] Set(“SIP/64.2.142.30-0884f098”, “MODE=unavail”) in new stack
– Executing [vmx@macro-vm:3] GotoIf(“SIP/64.2.142.30-0884f098”, “1?notdirect”) in new stack
– Goto (macro-vm,vmx,5)
– Executing [vmx@macro-vm:5] NoOp(“SIP/64.2.142.30-0884f098”, "e[1;35; 40mChecking if ext 501 is enabled: ") in new stack
– Executing [vmx@macro-vm:6] GotoIf(“SIP/64.2.142.30-0884f098”, “1?s-CHANUNAVAIL|1”) in new stack
– Goto (macro-vm,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-vm:1] Macro(“SIP/64.2.142.30-0884f098”, “get-vmcontext|501”) in new stack
– Executing [s@macro-get-vmcontext:1] Set(“e[1;3 5;40mSIP/64.2.142.30-0884f098”, “VMCONTEXT=default”) in new stack
– Executing [s@macro-get-vmcontext:2] GotoIf(“SIP/64.2.142.30-0884f098”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing [s@macro-get-vmcontext:300] NoOp(“SIP/64.2.142.30-0884f098”, “”) in new stack
– Executing [s-CHANUNAVAIL@macro-vm:2] e[1;36;40 mVoiceMail(“SIP/64.2.142.30-0884f098”, “501@default|su”) in new stack
Audio is at 69.64.59.250 port 19214
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 64.2.142.30:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK28597977;received=64.2.142.30;rport=5060
From: “GSA/FTS/6TTP AT” sip:[email protected];tag=as10e7067f
To: sip:[email protected];tag=as02ef4cba
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 3032 3032 IN IP4 69.64.59.250
s=session
c=IN IP4 69.64.59.250
t=0 0
m=audio 19214 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– <SIP/64.2.142.30-0884f098> Playing ‘/var/spool/asterisk/voicemail/default/501/unavail’ (language ‘en’)
pbx*CLI>
<— SIP read from 64.2.142.30:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK19eda7c7;rport
From: “GSA/FTS/6TTP AT” sip:[email protected];tag=as10e7067f
To: sip:[email protected];tag=as02ef4cba
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

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

<— SIP read from 64.2.142.30:5060 —>
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK32c892a3;rport
From: “GSA/FTS/6TTP AT” sip:[email protected];tag=as10e7067f
To: sip:[email protected];tag=as02ef4cba
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 64.2.142.30 : 5060 (NAT)

<— Transmitting (NAT) to 64.2.142.30:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK32c892a3;received=64.2.142.30;rport=5060
From: “GSA/FTS/6TTP AT” sip:[email protected];tag=as10e7067f
To: sip:[email protected];tag=as02ef4cba
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘SIP/64.2.142.30-0884f098’ in macro ‘vm’
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘SIP/64.2.142.30-0884f098’ in macro ‘exten-vm’
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/64.2.142.30-0884f098’
Really destroying SIP dialog ‘[email protected]’ Method: BYE
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER

2

quickly scanning the output it looks like none of your phone are registered. Which will give you exactly what you are describing.

at the asterisk cli type sip show peers and you will get a table like output.

output will look something like this:

Name/username              Host            Dyn Nat ACL Port     Status    
789/789                    172.18.1.89      D   N      5060     Unmonitored
420/420                    (Unspecified)    D   N      0        UNKNOWN   
252/252                    172.20.1.160     D   N      5060     OK (85 ms)

If yours are looking like extension 420 then that would be the issue as 420 is not registered on my system current.

3

I figured it out. I installed fail2ban and for some reason it was blocking my IP … even though I told it to ignore it.

thanks

4

ah, that would do it and an important detail that was not mentioned in your initial post.

5

yes… well, you would figure that the “ignore” statement in the config would actually work.

6