[Asterisk-Dev] Sip Hard Hungup

[James O. Sizemore III]

I Need to get the Hangup app to be
a true "hard hangup" when using sip.
The proxy software I use sends a
retransmit request on hangup. So
hangups always fails with Asterisk.
Example below.

What I want it to do is on receiving a
Hangup or a far end "BYE" all RTP is
canceled regardless of whether Asterisk
can find the call leg or not. A short
time out would be fine (to give Asterisk
compatible sip devices enough time to
finish a true sip hangup.) But if the
far end and Asterisk are incomputable
the RTP will stop eating up bandwidth.
And 4 second long voice mails will
not be left in the voice mail app on
calls that "did" hangup before entering
the voice mail app.

I am not terrible familiar with the
Asterisk code yet, I'm looking
through channels/chan_sip.c seeing
if I can get my head around it.

I guess "ast_rtp_destroy(p->rtp);"
need to find a home some place. LOL

Any pointer or thoughts?

-- Executing Goto("SIP/207.65.117.4:5061", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Hangup("SIP/207.65.117.4:5061", "") in new stack
== Spawn extension (default, #, 1) exited non-zero on 
'SIP/207.65.117.4:5061'
XXX Need to handle Retransmitting XXX:
BYE sip:2114 at 207.65.117.4 SIP/2.0
Via: SIP/2.0/UDP 207.65.117.8:5063;branch=5dcfa753
From: <sip:8500 at 207.65.117.4>;tag=05f38856
To: "324-2114"<sip:2114 at 207.65.117.4>;tag=003094c274b300327377afb9-73c7a81a
Call-ID: 003094c2-74b300ee-0671c7bd-6b35d109 at 207.65.7.178
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

  (no NAT) to 207.65.117.4:5061
Sip read: I>
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 207.65.117.8:5063;branch=5dcfa753
To: "324-2114"<sip:2114 at 207.65.117.4>;tag=003094c274b300327377afb9-73c7a81a
From: <sip:8500 at 207.65.117.4>;tag=05f38856
Call-ID: 003094c2-74b300ee-0671c7bd-6b35d109 at 207.65.7.178
CSeq: 102 BYE
Content-Length: 0


7 headers, 0 lines
Message is BYE