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

[Joshua C. Colp]

On Sat, Nov 16, 2019 at 7:59 PM Sebastian Nielsen <sebastian at sebbe.eu>
wrote:

> What would be the best way to solve this problem? Anyone else that have
> got the same problem with Android’s native SIP client, especially on
> Samsung phones?
>
>
>
> I do not know if the bug is in Android native SIP, or Samsung’s build of
> the SIP client, or if the bug is even with the OpenVPN client, or where the
> bug actually is.
>
> The ACK might even be sent for real, but have the incorrect source IP so
> asterisk ignores it.
>

The ACK is sent to the Contact header of the 200 OK sent to the phone.
Using the respective logging (sip set debug on or pjsip set logger on)
would tell you the IP address and port that Asterisk is telling the phone
to send to, and isolate the problem further. Asterisk also doesn't ignore
the ACK based on source IP address. If it shows up at Asterisk, it'll get
processed.


>
>
> Since audio works in both directions, it seems that the lack of ACK
> wouldn’t hurt (other than asterisk forcefully disconnecting the call) so I
> need to just tell Asterisk to not forcefully disconnect the callee.
>

Without modifying code there's no way. The 200 OK retransmits until it
gives up, and the call is disconnected.

-- 
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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