this problem came up when i tried forwarding calls..
-- Executing [1001@users:1] Macro("SIP/to_freepbx-0000003a", "stduser,1001,tT") in new stack -- Executing [s@macro-stduser:1] GotoIf("SIP/to_freepbx-0000003a", "1?FORWARD") in new stack -- Goto (macro-stduser,s,4) -- Executing [s@macro-stduser:4] Answer("SIP/to_freepbx-0000003a", "") in new stack -- Executing [s@macro-stduser:5] Goto("SIP/to_freepbx-0000003a", "users,1002,1") in new stack -- Goto (users,1002,1) == Channel 'SIP/to_freepbx-0000003a' jumping out of macro 'stduser' -- Executing [1002@users:1] Macro("SIP/to_freepbx-0000003a", "stduser,1002,tT") in new stack -- Executing [s@macro-stduser:1] GotoIf("SIP/to_freepbx-0000003a", "1?FORWARD") in new stack -- Goto (macro-stduser,s,4) -- Executing [s@macro-stduser:4] Answer("SIP/to_freepbx-0000003a", "") in new stack -- Executing [s@macro-stduser:5] Goto("SIP/to_freepbx-0000003a", "users,2004,1") in new stack -- Goto (users,2004,1) == Channel 'SIP/to_freepbx-0000003a' jumping out of macro 'stduser' -- Executing [2004@users:1] Dial("SIP/to_freepbx-0000003a", "SIP/2004@to_freepbx") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/2004@to_freepbx [Sep 8 12:24:54] NOTICE[17431]: chan_sip.c:21050 handle_response_invite: Failed to authenticate on INVITE to '"LEO" ;tag=as6388ac84' -- SIP/to_freepbx-0000003b is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/to_freepbx-0000003a' status is 'CONGESTION'
there seem to be no problem in the flow as seen on the logs except for the notification " chan_sip.c:21050 handle_response_invite: " Failed to authenticate on INVITE to "
i have two pbx servers.. one is gui-less asterisk while the other one is freepbx.. i created a sip trunk for them to connect..here it is
[general]
context=users
realm=training.com
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=gsm
language=en
trustrpid=yes
sendrpid=yes
[examconfig](!)
type=friend
host=dynamic
secret=1qaz1qaz
qualify=yes
callgroup=1
pickupgroup=1
context=users
canreinvite=no
[1001](examconfig)
mailbox=1001@default
callerid="Michael Jordan" <1001>
setvar=USERID=1001
[1002](examconfig)
mailbox=1002@default
callerid="Kobe Brian" <1002>
setvar=USERID=1002
[to_freepbx]
type=friend
host=192.168.1.250
insecure=port,invite
qualify=yes
context=users
disallow=all
allow=ulaw
allow=gsm
canreinvite=no
nat=no
dtmfmode=inband
here is a part my extensions.conf
enter code here
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
extenpatternmatchnew=no
[globals]
[users]
exten => _1XXX,1,Macro(stduser,${EXTEN},tT)
exten => _2XXX,1,Dial(SIP/${EXTEN}@to_freepbx)
exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@to_freepbx)
exten => _09X.,1,Dial(SIP/${EXTEN}@to_freepbx)
exten => 5002,1,GotoIftime(8:30-18:30,mon-fri,*,*?menu,s,1:menu_night,s,1)
include => features
[macro-stduser]
exten => s,1,GotoIf($["${DB(users/${ARG1}/FWD/Status)}" = "1"]?FORWARD)
exten => s,n,Dial(SIP/${ARG1},20)
exten => s,n,GotoIf($[“${DIALSTATUS}” = “NOANSWER”]?TIMEOUT)
exten => s,n(FORWARD),Answer()
exten => s,n,Goto(users,${DB(users/${ARG1}/FWD/Number)},1)
exten => s,n(TIMEOUT),Answer()
exten => s,n,Wait(1)
exten => s,n,Voicemail(${MACRO_EXTEN}@default,u)
exten => s,n,Hangup()
exten => h,1,NoOp(Shucks,hung up!)
when i enabled forwarding and tried calling from my local devices in asterisk, forwading is succesful
but when i try to call from freepbx to my asterisk local extension, it would go to congestion.. how do i troubleshoot this one
