[asterisk-users] Issue with handling of 480 DND

[Matt Fredrickson]

Response inline.

On Fri, Jan 6, 2017 at 12:47 PM, Markus Weiler <markus_weiler at mailworks.org>
wrote:

> Nobody any idea?
>
> It would be really helpful,
>
> Markus
>
>
>
>
> Am 06.01.2017 um 12:07 schrieb Markus Weiler:
>
>> Hi List,
>>
>> we're calling a sip phone from our Asterisk Server, and try to add logic
>> depending on the dialstatus
>>
>> Stripped down example;
>>
>> exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w)
>> exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1)
>> exten = 494XXXXXXXXX,n,Hangup()
>>
>>
>> .....
>> exten = 98-BUSY,1,NoOp(Busy)
>> exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
>> "1"]?Playback(/home/4120/mitarbeiter/ab))
>> ....
>> exten = 98-NOANSWER,1,NoOp(noanswer)
>> exten = 98-NOANSWER,n,ExecIf($["${Voicemail}" =
>> "1"]?Playback(/home/4120/mitarbeiter/ab))
>> ....
>>
>> Íf the phone call times out, the call is sent to 98-NOANSWER and then
>> answered as expected.
>> If the User presses DND on his phone the call is sent to 98-BUSY which
>> is identical but then the call is hung up. This behaviour is
>> unexpected/unwanted.
>>
>> We tried to figure out what the difference is and think it's how
>> Asterisk handles the "480 Do Not Disturb" from the phone
>> (xxx.xxx.xxx.xxx).
>> It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
>> is being forwarded".
>>
>> Is this a bug or a feature? :-) How could we handle this correctly?
>>
>
I believe that this is a consequence of the fact that when chan_sip
receives a "480 Do Not Disturb" it also queues a redirecting_update frame
on the channel. My guess is that the redirecting update is probably
triggering the "181 Call is being forward as well".  If you add the 'I'
flag to your dial, I believe it should suppress bridging of the redirection
information and I would think that would also cause the 181 not to be
sent.  If it's not that flag, I'd check the documentation for app_dial, as
I'm pretty sure there's a flag that should suppress that redirection from
from being bridged to the calling channel.

Hope that helps.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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