I have * Now Beta 5, connecting to Engin an Australian VoIP Provider.
When i make a call from a SIP phone to an australian landline number (via SIP trunk), if the end user hangs up before me, i here hold music and * keeps my channel open.
Interestingly if i call a mobile, * terminates both sessions correctly.
On another server which was running 1.2.13 with no problems…i upgraded to 1.4.2 (keeping the same config files), and it started not hanging up also…
is there a bug 1.4.2 or some SIP parameter i can play with for hangup detection?
[general]
context = default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap = no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
; Default is enabled
;realm=mydomain.tld bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = no
allowexternaldomains = no
allowexternalinvites = no
allowguest = yes
allowsubscribe = no
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
compactheaders = no
dumphistory = no
g726nonstandard = no
ignoreregexpire = no
jbenable = no
jbforce = no
jblog = no
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
notifyringing = no
pedantic = no
promiscredir = no
recordhistory = no
relaxdtmf = no
rtcachefriends = no
rtsavesysname = no
rtupdate = no
sendrpid = no
sipdebug = no
t1min = 100
t38pt_udptl = no
progressinband = no
trustrpid = no
usereqphone = no
videosupport = no
nat = no
####EXTENSIONS.CONF#####
in testing i just used:
exten =>600,1,Dial(SIP/${VOIP}/93812048)
and it wouldnt hang up
could it be something to with invalid BYE requests…i remember seeing something in the console about ‘invalid BYE request, not authenticated’ or something…at the time i saw it i wasnt making a call but maybe it was delayed or something??