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Summary:ASTERISK-25036: Asterisk still send an INVITE request after a call was canceled(realtime, rtcachefriends is enabled)
Reporter:Mikhail Fast (mouseratt1)Labels:
Date Opened:2015-04-30 03:32:38Date Closed:
Priority:MinorRegression?
Status:Open/NewComponents:Channels/chan_sip/General
Versions:13.0.0 13.1.0 13.2.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Debian wheezy x64Attachments:( 0) ASTERISK-25036.dump
( 1) debug-ASTERISK-25036.log
( 2) full.gz
( 3) sip-conf-ASTERISK-25036.txt
Description:Asterisk 13.2

Realtime, rtcachefriends=yes

Steps to reproduce a bug:

0) Asterisk has a public IP, cliens A and B connect over  NAT.
1) Regiser user A
2) Register user B
3) Unexpectedly disconnect client B (you may kill a proocess or simply disconnect a network cable)
4) Make a call from A to B(B exists in realtime cache, but really it is offline!)
5) Cancel a call
6) Register with a client B again

On secondary register you get a call into your sip-client B. In the console you may see, that asterisk still send an INVITE,  although a call already was canceled.
Comments:By: Mikhail Fast (mouseratt1) 2015-04-30 03:49:52.138-0500

sip.conf:
context from_anon_sip
allowguest  no
allowoverlap    no
udpbindaddr 0.0.0.0
transport   udp,ws,wss
tcpenable   no
tcpbindaddr 0.0.0.0
tlsenable   no
srvlookup   yes
disallow    all
allow   ulaw
allow   alaw
allow   g729
allow   g723
allow   gsm
allow   speex
trustrpid   no
sendrpid    rpid
dtmfmode    rfc2833
videosupport    no
maxcallbitrate  384
alwaysauthreject    yes
rtptimeout  60
rtpholdtimeout  300
allowsubscribe  no
t38pt_udptl yes,redundancy,maxdatagram=1024
nat force_rport,comedia
directmedia no
directrtpsetup  no
rtcachefriends  yes
rtsavesysname   yes
rtupdate    yes
rtautoclear 3600
ignoreregexpire no
progressinband  never
callcounter yes
use_q850_reason no
defaultexpiry   60
maxexpiry   60
By: Joshua C. Colp (jcolp) 2015-04-30 06:15:08.214-0500

Thanks for the report and debug. However we also need protocol specific debug captured at the time of the issue. Please include the following:

* Asterisk log files generated using the instructions on the Asterisk wiki [1], with the appropriate protocol debug options enabled, e.g. 'pjsip set logger on' if the issue involves the chan_pjsip channel driver.
* Configuration information for the relevant channel driver, e.g. pjsip.conf.
* A wireshark compatible packet capture, captured at the same time as the Asterisk log output.

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


By: Mikhail Fast (mouseratt1) 2015-05-13 07:07:36.798-0500

This is log of SIP trafic from asterisk in time of a call
By: Mikhail Fast (mouseratt1) 2015-05-13 07:09:11.734-0500

I' ve uploaded SIP log from asterisk. Is this enough?
By: Kirill Marchuk (62mkv) 2015-05-13 07:28:53.033-0500

by "this is a log of SIP traffic" Mikhail means https://issues.asterisk.org/jira/secure/attachment/52412/debug-ASTERISK-25036.log (IMHO)
By: Mikhail Fast (mouseratti) 2015-05-17 21:33:31.673-0500

I'm sorry, but what should I do now?
By: Rusty Newton (rnewton) 2015-05-19 19:37:33.334-0500

[~mouseratt1] If an issue is in Triage, you have to wait until a bug marshal is available to triage the issue. :)

Can you follow the https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information guide and provide a full log including the "DEBUG" and "VERBOSE" logs both turned up to 5 or above?

If possible, also include a packet capture that can be analyzed in wire-shark (tcpdump is probably the easiest way to gather one).

Thanks!
By: Rusty Newton (rnewton) 2015-05-19 19:37:54.772-0500

Use "Send Back" or "Enter Feedback" to send the issue back.
By: Rusty Newton (rnewton) 2015-06-03 15:00:18.025-0500

Mikhail will you be able to provide the information requested?

I'm unable to reproduce and can't go further without the debug and packet capture.
By: Mikhail Fast (mouseratti) 2015-06-16 07:28:43.284-0500

Full asterisk log in a moment of call
By: Mikhail Fast (mouseratti) 2015-06-16 07:35:16.103-0500

I've uploaded full.gz log and ASTERISK-25036.dump tcpdump file
By: Rusty Newton (rnewton) 2015-06-24 18:00:58.816-0500

reattaching reporter's sip peer configuration as .txt