Dear All,
I am a newbie, I have been hearing about Asterisk long time and interesting on it. I have installed Asterisk with A2billing.
I can set up trunk to My sip server and call can connect well. But i have try to add the secondary trunk with Ipbase authentication but my call cannot got thru. Please advice!
Below is my sip and extension.conf
[general]
context=a2billing ; Default context for incoming calls
;allowoverlap=no
allowguest=no
realm=203.223.46.56
;udpbindaddr=0.0.0.0
;tcpenable=no
;tcpbindaddr=0.0.0.0
bindaddr = 0.0.0.0
bindport=5060
srvlookup=yes
;videosupport = yes ; Enable video
disallow=all ; First disallow all codecs
allow = ulaw ; Allow codecs in order of preference
allow = 729
;allow = gsm
allow = alaw
;allow = h263 ; H.263 is our video codec
;allow = h263p ; H.263p is the enhanced video codec
dtmfmode=rfc2833 ; Set default dtmfmode for sending DTMF.
canreinvite=no
nat=yes
;qualify=yes
;insecure=port
rtcachefriends=yes
;rtupdate=no
;allowguest=no
;alwaysauthreject=yes
;defaultexpiry=90
externip = 203.223.46.55
register => 852123456:169169@203.223.46.56/852123456
[SIPNexge]
;context=a2billing
type=peer
;username=852123456
defaultuser=852123456
fromuser=852123456
secret=169169
host=203.223.46.56
;directmedia=no
;callerid= 852123456
;nat=yes
;qualify=yes
;canreinvite=no
;insecure=port, invite
;dtmfmode = rfc2833
;rtupdate=no
;port=5060
;externip=203.223.46.56
;_register=yes
[direct2gw7]
;disallow = all
;allow = alaw
;allow = 729
canreinvite = no
dtmfmode = rfc2833
host = 203.223.46.55
;insecure =very
;nat = yes
port = 5060
;qualify = yes
type = peer
My extension.conf
[general]
static=yes
writeprotect=yes
autofallthrough=yes
clearglobalvars=no
priorityjumping=yes
[a2billing]
; CallingCard application
;exten => _X.,1,NoOp(A2Billing Start)
;exten => _X.,n,DeadAgi(a2billing.php,1)
;exten => _X.,n,Hangup
;exten => _X.,1, DeadAGI(a2billing.php,1)
;exten => _X.,n,Wait()
;exten => _X.,n,Hangup()
;[a2billing]
exten => _X.,1,Answer
exten => _X.,2,Wait(1)
exten => _X.,3,Agi(a2billing.php,1)
exten => _X.,4,Wait(2)
exten => _X.,5,Hangup()
exten => 225,1,NoOp(A2Billing 2 Start)
exten => 225,n,Agi(a2billing.php,2)
exten => 225,n,Hangup()
;exten => 225,1,Answer(A2Billing 2 Start)
;exten => 225,2,Wait(1)
;exten => 225,1,Agi(a2billing.php,2)
;exten => 225,4,Wait(2)
;exten => 225,n,Hangup
;exten => _X.,1,Answer
;exten => _X.,1,Dial(SIP/sip-out/${EXTEN})
;exten => _X.,n,Hangup
[direct2gw7]
exten => _9X.,1,Dial(${direct2gw7}/${EXTEN:1})
exten => _9X.,n,Playtones(congestion)
exten => _9X.,n,Hangup()
[a2billing-did]
exten => _X.,1,Agi(a2billing.php,1,did)
exten => _X.,2,Hangup
[a2billing-callback]
exten => _X.,1,AGI(a2billing.php,1,callback)
exten => _X.,n,Hangup
extensions.conf (END)
blow is peer status and log summary.
localhost*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status Realtime
0151703678/0151703678 (Unspecified) D N 0 Unmonitored
1001/1001 (Unspecified) D N 0 Unmonitored
1002/1002 (Unspecified) D N 0 Unmonitored
1842784016/1842784016 192.168.0.185 D N 7657 Unmonitored
SIPNexge/852123456 119.82.248.16 N 5060 Unmonitored
direct2gw7 119.82.250.12 N 5060 Unmonitored
6 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 3 offline]
== Using SIP RTP CoS mark 5
– Executing [985512809606@a2billing:1] Answer(“SIP/1842784016-0000000f”, “”) in new stack
– Executing [985512809606@a2billing:2] Wait(“SIP/1842784016-0000000f”, “1”) in new stack
– Executing [985512809606@a2billing:3] AGI(“SIP/1842784016-0000000f”, “a2billing.php,1”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
– <SIP/1842784016-0000000f>AGI Script a2billing.php completed, returning 4
== Spawn extension (a2billing, 985512809606, 3) exited non-zero on ‘SIP/1842784016-0000000f’
Thanks and Regards
Danet