[asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer

[Joshua C. Colp]

On Sat, Nov 16, 2019 at 7:45 PM Sebastian Nielsen <sebastian at sebbe.eu>
wrote:

> Hello.
>
> I have a problem with the native Android SIP client, not acknowledging the
> call.
>
>
>
> Sent a message to the list for some weeks ago containing a sip debug log,
> but it only got stuck in moderation queue due to too large size (and it
> said I would get a message if moderators rejected it, but did not get
> message and I don’t think it got posted to list either)
>
>
>
> This ONLY happens when calling outgoing from the Android SIP client.
> Incoming calls works flawlessly.
>
>
>
> Everything works, audio in both directions, but the call is dropped after
> 30 sec.
>
> I have debugged it very much, and it seems that either Android is sending
> the acknowledge of the call to the incorrect IP (perhaps to the 3G network
> instead of via the VPN), or not sending it at all.
>
>
>
> BUT – Everything else is working flawlessly, including audio in both
> directions.
>
>
>
> So this means, I need somehow to tell Asterisk to ignore the lack of
> acknowledgement.
>
>
>
>
>
>
>
> So now to the question, since the call is dropped automatically after 30
> sec with ”NO_USER_RESPONSE” (Hangupcause 18) on the far end (the callee’s
> end), propably because the Android native Client is not acknowledging the
> connected call , is it possible to tell Asterisk to just ignore the lack of
> acknowledgement from Android somehow?
>
>
>
> Basically, for Client sip09 (username), never hang up for the reason 18
> (NO_USER_RESPONSE), threat like user response was received always.
>

There is no ability to ignore the lack of an ACK, as that violates the SIP
standard itself.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.sangoma.com & www.asterisk.org
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