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

[Sebastian Nielsen]

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.

 

 

Best regards, Sebastian Nielsen

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