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

[Steve Langstaff]

I don't know about the 1.4 source, but in 1.2 I guess you would have to
add some more code to

handle_response_peerpoke()

to handle the case where you got a 486 response from the peer.

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vieri
> Sent: 13 September 2007 18:56
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] how to determine if a SIP 
> extension has DNDonoroff
> 
> 
> --- 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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