BLF troubles - phones not getting updates

Hey guys,

Just set up a new asterisk box on site… all the phones are local to the box… calls in and out are working fine… but I can’t seem to get the BLFs to work… when I look in the CLI it looks to me like they’re set up correctly… but the phones don’t seem to be getting any changes in status… the lights stay green all the time no matter what…

Here’s what I see in the CLI –

-= Registered Asterisk Dial Plan Hints =-

104@from-internal : SIP/104 State:Idle Presence:not_set Watchers 15
104@jw-closed-ivr : SIP/104 State:Idle Presence:not_set Watchers 0
105@from-internal : SIP/105 State:Idle Presence:not_set Watchers 15
105@jw-closed-ivr : SIP/105 State:Idle Presence:not_set Watchers 0
106@from-internal : SIP/106 State:Idle Presence:not_set Watchers 15
106@jw-closed-ivr : SIP/106 State:Idle Presence:not_set Watchers 0
107@from-internal : SIP/107 State:Idle Presence:not_set Watchers 15
107@jw-closed-ivr : SIP/107 State:Idle Presence:not_set Watchers 0
100@from-internal : SIP/100 State:Idle Presence:not_set Watchers 15
100@jw-closed-ivr : SIP/100 State:Idle Presence:not_set Watchers 0
101@from-internal : SIP/101 State:Idle Presence:not_set Watchers 15
101@jw-closed-ivr : SIP/101 State:Idle Presence:not_set Watchers 0
102@from-internal : SIP/102 State:Idle Presence:not_set Watchers 15
102@jw-closed-ivr : SIP/102 State:Idle Presence:not_set Watchers 0
103@from-internal : SIP/103 State:Idle Presence:not_set Watchers 15
103@jw-closed-ivr : SIP/103 State:Idle Presence:not_set Watchers 0
108@from-internal : SIP/108 State:Idle Presence:not_set Watchers 15
108@jw-closed-ivr : SIP/108 State:Idle Presence:not_set Watchers 0
109@from-internal : SIP/109 State:Idle Presence:not_set Watchers 15
109@jw-closed-ivr : SIP/109 State:Idle Presence:not_set Watchers 0

Any ideas on where I should start looking ? The BLF on these phones was working fine when I had them connected to a server remotely through VPN… but now locally they don’t seem to like it…

Any help greatly appreciated !

Richard

Maybe I should also share how I have them set up in the extensions.conf —

exten=>100,hint,SIP/100
exten=>100,1,Dial(SIP/100,20,r)
exten=>100,2,Voicemail(100@default,u)
exten=>100,3,Hangup

exten=>101,hint,SIP/101
exten=>101,1,Dial(SIP/101,20,r)
exten=>101,2,Hangup

exten=>102,hint,SIP/102
exten=>102,1,Dial(SIP/102,20,r)
exten=>102,2,Voicemail(102@default,u)
exten=>102,3,Hangup

exten=>103,hint,SIP/103
exten=>103,1,Dial(SIP/103,20,r)
exten=>103,2,Voicemail(103@default,u)
exten=>103,3,Hangup

exten=>104,hint,SIP/104
exten=>104,1,Dial(SIP/104,20,r)
exten=>104,2,Voicemail(104@default,u)
exten=>104,3,Hangup

exten=>105,hint,SIP/105
exten=>105,1,Dial(SIP/105,20,r)
exten=>105,2,Voicemail(105@default,u)
exten=>105,3,Hangup

You could look at the SIP messages (e.g. pjsip set logger on) and identify the SUBSCRIBE/NOTIFY events to be sure that the phones get the events. Then you should check, whether the phones are properly configured.

I am not quite sure what to look for as far as the subscribe messages go for the BLFs… I found this… Does this mean the feature is working ?

<— SIP read from UDP:192.168.5.119:5060 —>
SUBSCRIBE sip:asterisk@192.168.5.7:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.119:5060;branch=z9hG4bK2096163203;rport
From: sip:109@192.168.5.7;tag=765343929
To: sip:109@192.168.5.7;tag=as3a6ac999
Call-ID: 244750482-5060-1256@BJC.BGI.F.BBJ
CSeq: 20300 SUBSCRIBE
Contact: sip:109@192.168.5.119:5060
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.11.3
Expires: 1800
Supported: replaces, path, timer
Event: message-summary
Accept: application/simple-message-summary
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Found peer ‘109’ for ‘109’ from 192.168.5.119:5060

<— Transmitting (no NAT) to 192.168.5.119:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.5.119:5060;branch=z9hG4bK2096163203;received=192.168.5.119;rport=5060
From: sip:109@192.168.5.7;tag=765343929
To: sip:109@192.168.5.7;tag=as3a6ac999
Call-ID: 244750482-5060-1256@BJC.BGI.F.BBJ
CSeq: 20300 SUBSCRIBE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“19857e96”
Content-Length: 0

You haven’t configured a password in the phones, or you haven’t provided the full logs.

The logs are huge… it would help to know what I’m looking for as far as the BLFs hints / subscriptions… any ideas what I would search for ?

A subscribe with something other than a 401 response. Every first attempt, where a password is involved, will generate a 401 response. If the phone has a password properly configured, it will repeat the request with authentication data. If it doesn’t repeat like that, the phone doesn’t have a password set, but Asterisk does.

If the password was incorrect… wouldn’t the phones not be able to call out or receive calls ? Because the phones are working fine as far as that goes… just the BLFs aren’t working…

I’m not talking about incorrect passwords, but rather about no password at all. It the password is incorrect, you will get SUBSCRIBE - 401 - SUBSCRIBE - 403 or 603.

If you are only getting 401, it means the phones does not know what password to use.

If the phones can register, or initiate calls, they probably do have passwords set, and the problem is probably that you didn’t provide the complete log for the transaction.

Note most phones can receive calls without passwords, although they would normally have had to have registered, for the PBX to know how to route the call to them.

Looks like this extension is showing that it is subscribed… I’ve just enabled full logs on the server so I will take a look at those in a bit after they’ve collected several calls… But nothing jumps out at me as to why the BLFs aren’t working…

  • Name : 100
    Description :
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-internal
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. :
    Language :
    Tonezone :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox : 100@default
    VM Extension : asterisk
    LastMsgsSent : 2/0
    Call limit : 0
    Max forwards : 0
    Busy level : 1
    Dynamic : Yes
    Callerid : “” <>
    MaxCallBR : 384 kbps
    Expire : 455
    Insecure : invite
    Force rport : Auto (No)
    Symmetric RTP: No
    ACL : No
    ContactACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: 4294967295
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    Path support : No
    Path : N/A
    TrustIDOutbnd: Legacy
    Subscriptions: Yes
    Overlap dial : No
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP : 192.168.5.113:5060
    Defaddr->IP : (null)
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username: 100
    SIP Options : (none)
    Codecs : (ulaw|alaw|gsm|h263|amr|amrwb|codec2|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263p|h264|mpeg4|vp8|vp9|red|t140|t38|silk|silk|silk|silk)
    Auto-Framing : No
    Status : OK (4 ms)
    Useragent : Grandstream GXP2160 1.0.11.3
    Reg. Contact : sip:100@192.168.5.113:5060
    Qualify Freq : 60000 ms
    Keepalive : 0 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : No
    RTCP Mux : No

Ok maybe I’m on to something here… on another Asterisk box that I have and that the BLFs ARE working… i see these messages going by –

But I do not see these on the box with BLFs not working… is there maybe a feature or function that might be OFF ?

Extension Changed 4[from-internal] new state Ringing for Notify User 2
== Extension Changed 4[from-internal] new state Ringing for Notify User 5
== Extension Changed 4[from-internal] new state Ringing for Notify User 1
== Extension Changed 4[from-internal] new state Ringing for Notify User 3
== Extension Changed 4[from-internal] new state Ringing for Notify User 4
– SIP/4-00009282 is ringing
== Extension Changed 4[from-internal] new state Ringing for Notify User 2
== Extension Changed 4[from-internal] new state Ringing for Notify User 5
== Extension Changed 4[from-internal] new state Ringing for Notify User 1
== Extension Changed 4[from-internal] new state Ringing for Notify User 3
== Extension Changed 4[from-internal] new state Ringing for Notify User 4
– Nobody picked up in 20000 ms
== Extension Changed 4[from-internal] new state Idle for Notify User 2
== Extension Changed 4[from-internal] new state Idle for Notify User 5
== Extension Changed 4[from-internal] new state Idle for Notify User 1
== Extension Changed 4[from-internal] new state Idle for Notify User 3
== Extension Changed 4[from-internal] new state Idle for Notify User 4

Perhaps the “callcounter” option in sip.conf?

While we’re at it, it couldn’t hurt to take a look:
https://wiki.asterisk.org/wiki/display/AST/Configuring+chan_sip

That was exactly it !!! bingo! :slight_smile: problem solved!

callcounter = yes

uncommented that in sip.conf… did a core reload… and poof! fixed!

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