[asterisk-bugs] [JIRA] (ASTERISK-25036) Asterisk still send an INVITE request after a call was canceled(realtime, rtcachefriends is enabled)

[Mikhail Fast (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25036?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Mikhail Fast updated ASTERISK-25036:
------------------------------------


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

> Asterisk still send an INVITE request after a call was canceled(realtime, rtcachefriends is enabled)
> ----------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-25036
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25036
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 13.0.0, 13.1.0, 13.2.0
>         Environment: Debian wheezy x64
>            Reporter: Mikhail Fast
>            Severity: Minor
>
> 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. 



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