Freepbx dropped all registrations unknown why

I stopped using my system out of fustration. Anyway, I would like to get it going again. It dropped my polycom ip500 registration “does not show up in the cli using show registration” and the sipura ata 1000. Where are the error logs in the freepbx so I can show them here?

I did or made no changes to my working system then one day, no phones were registered.

Thanks

Basic

Core 2.8.1.0 FreePBX Enabled  
DAHDi Config 2.7.0 Digium Enabled  
Digium Addons 2.7.0 Digium Enabled  
Fax Configuration 2.8.0.5 Schmoozecom.com Enabled  
Feature Code Admin 2.8.0.1 FreePBX Enabled  
FreePBX Framework 2.8.1.4 FreePBX Enabled  
SIPSTATION   Bandwidth.com Not Installed (Locally available)  
System Dashboard 2.7.0.1 FreePBX Enabled  
Voicemail 2.8.0.0 FreePBX Enabled  

Inbound Call Control

IVR 2.8.0.5 FreePBX Enabled  
Time Conditions 2.8.0.3 FreePBX Enabled  

Internal Options & Configuration

Conferences 2.8.0.2 FreePBX Enabled  
Info Services 2.7.0.0 FreePBX Enabled  
Music on Hold 2.8.0.3 FreePBX Enabled  
Paging and Intercom 2.8.0.1 FreePBX Enabled  
Recordings 3.3.10.3 FreePBX Enabled  

System Administration

Asterisk IAX Settings 2.8.0.0 Bandwidth.com Enabled  
Asterisk SIP Settings 2.7.0.1 Bandwidth.com Enabled  
Custom Applications 2.7.0.0 FreePBX Enable

From the asterisk command line, what do you get when you do:

sip show settings

and

sip set debug on

If no sip traffic appears, asterisk isn’t getting it.

[Apr 13 10:54:36] VERBOSE[3117] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
[Apr 13 10:54:36] VERBOSE[3117] manager.c: == Manager ‘admin’ logged off from 127.0.0.1
[[email protected] asterisk]# tail -30 full
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_general_additional.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_general_custom.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_nat.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_registrations_custom.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_registrations.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_custom.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_additional.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_custom_post.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] netsock.c: == Using SIP TOS bits 96
[Apr 13 11:03:01] VERBOSE[2909] netsock.c: == Using SIP CoS mark 4
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_notify.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_notify_custom.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[2909] config.c: == Parsing ‘/etc/asterisk/sip_notify_additional.conf’: [Apr 13 11:03:01] VERBOSE[2909] config.c: == Found
[Apr 13 11:03:01] VERBOSE[3275] loader.c: – Reloading module ‘res_crypto.so’ (Cryptographic Digital Signatures)
[Apr 13 11:03:01] VERBOSE[3275] loader.c: – Reloading module ‘chan_mgcp.so’ (Media Gateway Control Protocol (MGCP))
[Apr 13 11:03:01] VERBOSE[3275] loader.c: – Reloading module ‘app_queue.so’ (True Call Queueing)
[Apr 13 11:03:01] NOTICE[3275] app_queue.c: No queuerules.conf file found, queues will not follow penalty rules
[Apr 13 11:03:01] VERBOSE[3275] config.c: == Parsing ‘/etc/asterisk/queues.conf’: [Apr 13 11:03:01] VERBOSE[3275] config.c: == Found
[Apr 13 11:03:01] VERBOSE[3275] config.c: == Parsing ‘/etc/asterisk/queues_general_additional.conf’: [Apr 13 11:03:01] VERBOSE[3275] config.c: == Found
[Apr 13 11:03:01] VERBOSE[3275] config.c: == Parsing ‘/etc/asterisk/queues_custom_general.conf’: [Apr 13 11:03:01] VERBOSE[3275] config.c: == Found
[Apr 13 11:03:01] VERBOSE[3275] config.c: == Parsing ‘/etc/asterisk/queues_custom.conf’: [Apr 13 11:03:01] VERBOSE[3275] config.c: == Found
[Apr 13 11:03:01] VERBOSE[3275] config.c: == Parsing ‘/etc/asterisk/queues_additional.conf’: [Apr 13 11:03:01] VERBOSE[3275] config.c: == Found
[Apr 13 11:03:01] VERBOSE[3275] config.c: == Parsing ‘/etc/asterisk/queues_post_custom.conf’: [Apr 13 11:03:01] VERBOSE[3275] config.c: == Found
[Apr 13 11:03:01] VERBOSE[3275] loader.c: – Reloading module ‘res_clialiases.so’ (CLI Aliases)
[Apr 13 11:03:01] ERROR[3275] res_clialiases.c: res_clialiases configuration file ‘cli_aliases.conf’ not found
[Apr 13 11:03:01] VERBOSE[3275] loader.c: – Reloading module ‘codec_gsm.so’ (GSM Coder/Decoder)
[Apr 13 11:03:01] VERBOSE[2908] chan_mgcp.c: Reloading MGCP
[Apr 13 11:03:01] NOTICE[2908] chan_mgcp.c: Unable to load config mgcp.conf, MGCP disabled
[Apr 13 11:03:01] VERBOSE[3275] manager.c: == Manager ‘admin’ logged off from 127.0.0.1
[[email protected] asterisk]#

The only time this has happened to me is because of a network issue. Are you using DHCP with your phones or fixed IP addresses? If DHCP are the phones receiving an IP address via dhcp?

Can you ping the phones from the server?

Bill/W5WAF

I know you are probably thinking it was a physical or network layer issue. It is not. I can access the sipura 1000 ata web interface with no issues.

BTW, other then the /var/log/asterisk/full error log, is there one in the freepbx interface? I do not see one.

here is cli

sip show peers
Name/username Host Dyn Nat ACL Port Status
200 (Unspecified) D N A 5060 UNKNOWN
201 (Unspecified) D N A 5060 UNKNOWN
202 (Unspecified) D N A 5060 UNKNOWN
203 (Unspecified) D N A 5060 UNKNOWN
929 (Unspecified) D N A 5060 UNKNOWN
5 sip peers [Monitored: 0 online, 5 offline Unmonitored: 0 online, 0 offline

sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.

I might just add a software sip phone like xten again and see what is happening.

Update: I also now have a issue with no being able to add new extensions. I can fill in the blanks as always, click update, but the list of extensions do not add the new extension or does the orange button to reload the extension show up.

What is the likely cause in this issue?

I have sip debug on for at least two hours now. No activity on cli. If there is no sip phones registered, is this normal?

ping gets a reply from the asterisk server. Is my problem unique do to the fact I have had few replies?

I may just upgrade freepbx to see if that resolves the problem.

I also want to emphasis, freepbx WILL NOT add new extensions in the interface. I create it, update it, but it does not show up in the extensions list on the right of the screen. This problem is more then likely tied to this issue.

Hi skykingOH,

No idea what is going on. To many other users here mum or dont show interest in this unusual problem. Any ideas?

Connected to Asterisk 1.6.2.15 currently running on localhost (pid = 2790)
Verbosity is at least 3
– Remote UNIX connection
localhost*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
200 (Unspecified) D N A 5060 UNKNOWN
201 (Unspecified) D N A 5060 UNKNOWN
202 (Unspecified) D N A 5060 UNKNOWN
203 (Unspecified) D N A 5060 UNKNOWN
929 (Unspecified) D N A 5060 UNKNOWN

I am concerned with extentions 200 and 201

localhost*CLI> sip show peer 200

  • Name : 200
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-internal
    Subscr.Cont. :
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : [email protected]
    VM Extension : *97
    LastMsgsSent : 32767/65535
    Call limit : 2147483647
    Dynamic : Yes
    Callerid : “device” <200>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Nat : Always
    ACL : Yes
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: -1
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    Forward Loop : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP : (Unspecified) Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username:
    SIP Options : (none)
    Codecs : 0xa8e (gsm|ulaw|alaw|g726|lpc10|speex)
    Codec Order : (ulaw:20,alaw:20,gsm:20,lpc10:20,speex:20,g726:20)
    Auto-Framing : No
    100 on REG : No
    Status : UNKNOWN
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    Parkinglot :

localhost*CLI> sip show peer 201

  • Name : 201
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-internal
    Subscr.Cont. :
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : [email protected]
    VM Extension : *97
    LastMsgsSent : 32767/65535
    Call limit : 2147483647
    Dynamic : Yes
    Callerid : “device” <201>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Nat : Always
    ACL : Yes
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: -1
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    Forward Loop : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP : (Unspecified) Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username:
    SIP Options : (none)
    Codecs : 0xa8e (gsm|ulaw|alaw|g726|lpc10|speex)
    Codec Order : (ulaw:20,alaw:20,gsm:20,lpc10:20,speex:20,g726:20)
    Auto-Framing : No
    100 on REG : No
    Status : UNKNOWN
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    Parkinglot :

If you do a sip show peers in Asterisk does the peer display in the list? Then you can do a a ‘sip show peer xxx’ to see the details.

I have sip debug on, and getting no traffic. What next?

new-host-3*CLI> sip show settings

Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.9.0(1.8.4.2)
SDP Session Name: Asterisk PBX 1.8.4.2
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No

Network Settings:

SIP address remapping: Disabled, no localnet list
Externhost:
externaddr: (null)
Externrefresh: 10

Global Signalling Settings:

Codecs: 0xe (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Force rport: No
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97