[asterisk-users] how to determine if a SIP extension has DND onoroff

[Vieri]

--- Steve Langstaff <steve.langstaff at citel.com> wrote:

> Can you hook into the "qualify" code somehow? - that
> uses SIP OPTIONS.

I already knew of this wiki page:
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

So I did a "sip show peer" on the asterisk cli which I
am supposing is the same as the SIPPEER function.

When SIP softphone has DND turned OFF:

INF-VOIP*CLI> sip show peer 4053
INF-VOIP*CLI>

  * Name       : 4053
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-internal
  Subscr.Cont. : <Not set>
  Language     : es
  AMA flags    : Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Mailbox      : 4053 at device
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "device" <4053>
  Expire       : 58
  Insecure     : no
  Nat          : Always
  ACL          : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID    : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 10.215.147.240 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 4053
  SIP Options  : (none)
  Codecs       : 0xc (ulaw|alaw)
  Codec Order  : (ulaw,alaw)
  Status       : OK (127 ms)
  Useragent    : SJphone/1.65.377a (SJ Labs)
  Reg. Contact : sip:4053 at 10.215.147.240

When SIP softphone has DND turned ON:

INF-VOIP*CLI> sip show peer 4053
INF-VOIP*CLI>

  * Name       : 4053
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-internal
  Subscr.Cont. : <Not set>
  Language     : es
  AMA flags    : Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Mailbox      : 4053 at device
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "device" <4053>
  Expire       : 45
  Insecure     : no
  Nat          : Always
  ACL          : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID    : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 10.215.147.240 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 4053
  SIP Options  : (none)
  Codecs       : 0xc (ulaw|alaw)
  Codec Order  : (ulaw,alaw)
  Status       : OK (127 ms)
  Useragent    : SJphone/1.65.377a (SJ Labs)
  Reg. Contact : sip:4053 at 10.215.147.240
INF-VOIP*CLI>

I don't see any difference and "SIP Options  : (none)"
doesn't look "good".

(the SIP extension has qualify=yes)



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