[asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

[Mark Michelson]

Barry L. Kline wrote:
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> I must be missing something here but I can't figure out why I can't get
> DEVICE_STATE() to give me anything other than "NOT_INUSE".
> 
> I have two extensions:  6666 and 6668.  I used 6668 to make a call to
> yet another phone, so I know that it's busy.  I then use 6666 to call
> 6668 and in the dialplan have a noop to see what DEVICE_STATE() is
> returning for both extensions.
> 
> I get:
> 
> 
> [Jul 15 17:20:43]     -- Executing [6668 at sip-deskset:1]
> NoOp("SIP/6666-08636430", "SIP/6668 has state NOT_INUSE") in new stack
> [Jul 15 17:20:43]     -- Executing [6668 at sip-deskset:2]
> NoOp("SIP/6666-08636430", "SIP/6666 has state NOT_INUSE") in new stack
> 
> 
> 6668 is configure so that I get this:
> 
>   * Name       : 6668
>   Secret       : <Set>
>   MD5Secret    : <Not set>
>   Context      : sip-deskset
>   Subscr.Cont. : <Not set>
>   Language     :
>   AMA flags    : Unknown
>   Transfer mode: open
>   CallingPres  : Presentation Allowed, Not Screened
>   Callgroup    :
>   Pickupgroup  :
>   Mailbox      :
>   VM Extension : asterisk
>   LastMsgsSent : 32767/65535
>   Call limit   : 99
>   Busy level   : 1
>   Dynamic      : Yes
>   Callerid     : "Matts SIP" <6668>
>   MaxCallBR    : 384 kbps
>   Expire       : 2016
>   Insecure     : no
>   Nat          : RFC3581
>   ACL          : No
>   T38 pt UDPTL : No
>   CanReinvite  : No
>   PromiscRedir : No
>   User=Phone   : No
>   Video Support: No
>   Text Support : No
>   Trust RPID   : No
>   Send RPID    : No
>   Subscriptions: Yes
>   Overlap dial : No
>   DTMFmode     : rfc2833
>   Timer T1     : 500
>   Timer B      : 32000
>   ToHost       :
>   Addr->IP     : 192.168.1.70 Port 5060
>   Defaddr->IP  : 0.0.0.0 Port 5060
>   Transport    : UDP
>   Def. Username: Barry's IP450
>   SIP Options  : (none)
>   Codecs       : 0x4 (ulaw)
>   Codec Order  : (ulaw:20)
>   Auto-Framing :  No
>   100 on REG   : No
>   Status       : OK (14 ms)
>   Useragent    : PolycomSoundPointIP-SPIP_450-UA/3.1.3.0439
>   Reg. Contact : sip:6668 at 192.168.1.70
>   Qualify Freq : 60000 ms
>   Sess-Timers  : Accept
>   Sess-Refresh : uas
>   Sess-Expires : 1800 secs
>   Min-Sess     : 90 secs
> 
> 
> - From what I have read, with 'Busy Level = 1' I should be seeing BUSY
> returned from the DEVICE_STATE() call, yet I don't.
> 
> What is the super-secret sauce required to get Asterisk to return the
> correct state?
> 
> TIA,
> 
> Barry

You need to set a call-limit for the SIP peer. Device state calculation for a 
SIP peer is predicated on both the call-limit and busylevel. Let's say that you 
were to have a call-limit of 2, but no busylevel set. These are the device 
states reported for the peer based on the number of calls currently handled:

0 calls: not in use
1 call: in use
2 calls: busy

Basically, the busylevel defaults to the call-limit value. Now if you add a 
busylevel = 1 to sip.conf, these are the device states reported:

0 calls: not in use
1 call: busy
2 calls: busy

Mark Michelson