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Barry L. Kline

2009-Jul-15 21:26 UTC

[asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

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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
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Mark Michelson

2009-Jul-15 21:48 UTC

head link

[asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

Barry L. Kline wrote:> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> 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

Philipp Kempgen

2009-Jul-15 21:52 UTC

head link

[asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

Barry L. Kline schrieb:> 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.
> What is the super-secret sauce required to get Asterisk to return the
> correct state?
Just to be sure: Do you have hints configured for the extensions?
See http://das-asterisk-buch.de/2.1/blf-leds.html
(The text is in german but there are many examples in extensions.conf
and extensions.ael syntax. "Zur?ck" = "Previous",
"Weiter" = "Next")


    Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
--

Barry L. Kline

2009-Jul-15 22:46 UTC

head link

[asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

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Philipp Kempgen wrote:
> Just to be sure: Do you have hints configured for the extensions?
> See http://das-asterisk-buch.de/2.1/blf-leds.html
> (The text is in german but there are many examples in extensions.conf
> and extensions.ael syntax. "Zur?ck" = "Previous",
"Weiter" = "Next")
Hi Philipp.

I did configure the hints (see my other reply) and things are still not
working properly.  Thanks for your suggestion!

Barry

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Andrew Thomas

2009-Jul-16 08:28 UTC

head link

[asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

Why are you putting semi-colons at the end of every line?  The dialplan
isn't written in PHP ;).

	  	

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Barry L.
Kline
Sent: 15 July 2009 23:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

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Mark Michelson wrote:> 
> 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 currentlyhandled:


Hi Mark.  Thanks for your explanation of these parameters.

I should have posted my configurations.  I double-checked the contents
of sip.conf and I have this.  The 'subscribecontext' was added for
testing, per the other reply I got for my question.

;
; Settings common to all devices on our system
;
[basic-options](!)
        type=friend
        host=dynamic
        canreinvite=no
        disallow=all
        allow=ulaw
        dtmfmode=rfc2833
        qualify=yes

;
; Standard desksets here
;
[lan-deskset](!,basic-options)
        context=sip-deskset
        notifyringing = yes
        notifyhold = yes
        limitonpeers = yes
        call-limit=99

[6668](lan-deskset)
        secret=mysecret
        callerid="Matts SIP" <6668>
        username=Barry's IP450
	call-limit=32
        busylevel=1
        subscribecontext=hint-context


My hint-context is:

[hint-context]

exten => 6668,hint,SIP/6668;


I'm still not getting anything other than NOT_INUSE from DEVICE_STATE.
Here is the CLI output:

[Jul 15 18:40:15]     -- Executing [6668 at sip-deskset:1]
NoOp("SIP/6666-0955ecc8", "SIP/6668 has state NOT_INUSE") in
new stack
[Jul 15 18:40:15]     -- Executing [6668 at sip-deskset:2]
NoOp("SIP/6666-0955ecc8", "SIP/6666 has state NOT_INUSE") in
new stack
[Jul 15 18:40:15]     -- Executing [6668 at sip-deskset:3]
ExecIf("SIP/6666-0955ecc8", "0?Busy(10)") in new stack
[Jul 15 18:40:15]     -- Executing [6668 at sip-deskset:4]
Dial("SIP/6666-0955ecc8", "SIP/6668") in new stack


And here is sip show inuse:

corp-asterisk*CLI> sip show inuse
* User name               In use          Limit
6668                      1               32
6667                      0               99
6666                      1               99
* Peer name               In use          Limit
6668                      1/1/0           32
6667                      0/0/0           99
6666                      0/0/0           99


For completeness, here is the dialplan that's producing this:

exten => 6668,1,NoOp(SIP/${EXTEN} has state
${DEVICE_STATE(SIP/${EXTEN})});
exten => 6668,n,NoOp(SIP/6666 has state ${DEVICE_STATE(SIP/6666)});
exten =>
6668,n,ExecIf($[${DEVICE_STATE(SIP/${EXTEN})}="BUSY"]?Busy(10));
exten => 6668,n,Dial(SIP/${EXTEN});


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