provided by Star2Billing S.L.

Hi,

Please help me i have install a2billing 1.4 and asterisk 1.6 on my server everything is working as i check my PC to PC calls are running very smoothly. But when i try to male PC to PSTN call to any number i am geting the following error message.

Using SIP RTP CoS mark 5
-- Executing [00919811655464@adore:1] Dial("SIP/76719-092c1e10", "SIP/00919811655464") in new stack
== Using SIP RTP CoS mark 5
[Aug 20 12:23:50] WARNING[11363]: chan_sip.c:4224 create_addr: No such host: 00919811655464
[Aug 20 12:23:50] WARNING[11363]: app_dial.c:1468 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
[Aug 20 12:23:50] WARNING[11363]: pbx.c:3080 pbx_extension_helper: No application 'Hungup' for extension (adore, 00919811655464, 2)
== Spawn extension (adore, 00919811655464, 2) exited non-zero on 'SIP/76719-092c1e10'

This is my Sip.conf

[general]
context = default
realm = XXXX
bindport = 5060
bindaddr = XXXXXX
insecure = very
disallow = all
allow = g723
allow = g729
allow = gsm
allow = ulaw
allow = alaw
echocancel = yes
echocancelwhenbridged = yes
;jbenable = yes
dtmfmode = rfc2833
dtmfcompensate = yes
relaxdtmf = yes
externip = XXXXX
canreinvite = update
allowsubscribe = yes
trustrpid = no



[sip.voicetrading.com]

authuser=XXXXX
fromdomain=sip.voicetrading.com
fromuser=XXXXX
host=sip.voicetrading.com
insecure=very
nat=1
qualify=yes
secret=XXXX
type=peer
username=XXXX
disallow = all
allow = g729,ulaw
register => eurovoip2:[email protected]

My Extensions.conf :

[adore]
;################################################################
; PC-to-PC for SIP
exten => _X.,1,Dial(SIP/${EXTEN:0})
exten => _X.,n,Hungup()
;################################################################
; PC-to-PC for IAX2
exten => _*.,1,Dial(IAX2/${EXTEN:1})
exten => _*.,n,Hungup()
;################################################################
; PC-to-Phone for SIP & IAX2
exten => _X.,1,Answer()
;exten => _X.,n,Wait(2)
exten => _X.,n,DeadAGI(a2billing.php|2)
exten => _X.,n,Wait(2)
exten => _X.,n,Hangup()
;################################################################
;exten => s,1,Hungup()
;exten => s,n,Hungup()
;################################################################


Please help me to over come this.

Regards
Aryan Singh

Maybe your provider doesn't support the SIP INVITE messages sent by A2Billing. I had that problem.
So I deleted the string in A2Billing Admin panel -> System Settings -> Group List -> agi-conf1 -> Dial Command Params

After that it worked.

Be sure to backup the string before you delete it, just in case that is not the issue.

Hope I helped