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.
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
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?
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.
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.
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 : 200@default
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 : 201@default
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 :
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