asterisk users - Apr 2011 - Registrations stops after 403 FORBIDDEN

Hello list,

I have in sip.conf :


/maxexpiry=60                 ; Maximum allowed time of incoming 
registrations
                                 ; and subscriptions (seconds)
minexpiry=60                   ; Minimum length of 
registrations/subscriptions (default 60)
defaultexpiry=120              ; Default length of incoming/outgoing 
registration

;----------------------------------------- OUTBOUND SIP REGISTRATIONS  
------------------------
registertimeout=240             ; retry registration calls every 20 
seconds (default)
;registerattempts=0            ; Number of registration attempts before 
we give up
                                 ; 0 = continue forever, hammering the 
other server
                                 ; until it accepts the registration
                                 ; Default is 0 tries, continue forever/


When I register to my SIP provider and this provider sends a 403 
FORBIDDEN as SIP response, then Asterisk stops registering.

I thought that with "registerattempts=0" Asterisk will keep on sending
a
REGISTER untill the other ends accepts it ?

To be clear, my SIP provider sometimes rejects a registration but that's 
a bug in their system at the moment. So I need Asterisk to try multiple 
times until the providers's system accepts it. It happens often and at 
the moment I notice too late that the number is not registered to my SIP 
provider. While if Asterisk keeps on trying, the registration will 
finally come through again.


So are my settings wrong ?


Kind regards,
Jonas.
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On Mon, Apr 18, 2011 at 4:54 AM, Jonas Kellens <jonas.kellens at
telenet.be>wrote:
>  Hello list,
>
> I have in sip.conf :
>
<snip>

So are my settings wrong ?
>
>
What does sip show settings look like from the CLI?

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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On 04/18/2011 05:33 PM, Warren Selby wrote:
> On Mon, Apr 18, 2011 at 4:54 AM, Jonas Kellens 
> <jonas.kellens at telenet.be <mailto:jonas.kellens at
telenet.be>> wrote:
>
>     Hello list,
>
>     I have in sip.conf :
>
>
> <snip>
>
>     So are my settings wrong ?
>
>
> What does sip show settings look like from the CLI?
vps*CLI> sip show settings


Global Settings:
----------------
   UDP SIP Port:           5060
   UDP Bindaddress:        0.0.0.0
   TCP SIP Port:           Disabled
   TLS SIP Port:           Disabled
   Videosupport:           Yes
   Textsupport:            No
   Ignore SDP sess. ver.:  No
   AutoCreate Peer:        No
   Match Auth Username:    No
   Allow unknown access:   No
   Allow subscriptions:    Yes
   Allow overlap dialing:  No
   Allow promsic. redir:   No
   Enable call counters:   Yes
   SIP domain support:     No
   Realm. auth:            No
   Our auth realm          domain.be
   Call to non-local dom.: Yes
   URI user is phone no:   No
   Always auth rejects:    Yes
   Direct RTP setup:       No
   User Agent:             Asterisk PBX 1.6.2.16.1
   SDP Session Name:       Asterisk PBX 1.6.2.16.1
   SDP Owner Name:         owner
   Reg. context:           (not set)
   Regexten on Qualify:    No
   Caller ID:              0
   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:           Enabled
   Qualify Freq :          120000 ms

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:         3
   802.1p CoS RTP audio:   5
   802.1p CoS RTP video:   4
   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: <none>
   Externip:               0.0.0.0:0
   Externrefresh:          10
   STUN server:            0.0.0.0:0

Global Signalling Settings:
---------------------------
   Codecs:                 0x28090a (gsm|alaw|g726|g729|h263|h264)
   Codec Order:            alaw:20,g726:20,g729:20,gsm:20
   Relax DTMF:             No
   RFC2833 Compensation:   No
   Compact SIP headers:    No
   RTP Keepalive:          0 (Disabled)
   RTP Timeout:            60
   RTP Hold Timeout:       0 (Disabled)
   MWI NOTIFY mime type:   application/simple-message-summary
   DNS SRV lookup:         No
   Pedantic SIP support:   No
   Reg. min duration       60 secs
   Reg. max duration:      60 secs
   Reg. default duration:  300 secs
   Outbound reg. timeout:  240 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: <not set>
   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

Default Settings:
-----------------
   Allowed transports:     UDP
   Outbound transport:      UDP
   Context:                default
   Nat:                    RFC3581
   DTMF:                   rfc2833
   Qualify:                0
   Use ClientCode:         No
   Progress inband:        Never
   Language:               nl
   MOH Interpret:          default
   MOH Suggest:
   Voice Mail Extension:   asterisk
   Forward Detected Loops: Yes

Realtime SIP Settings:
----------------------
   Realtime Peers:         Yes
   Realtime Regs:          No
   Cache Friends:          Yes
   Update:                 Yes
   Ignore Reg. Expire:     No
   Save sys. name:         No
   Auto Clear:             120 (Disabled)

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