[Help] Asterisk - Callmanager calling problem


Hi everybody,

i’m trying to connect my asterisk to a Cisco Callmanager 4.1 using a SIP trunk. I’ve been following the instructions specified in voip-info.org/wiki/view/Aste … ntegration, but the only thing that i’ve obtained is the possibility of make calls between an asterisk extension and a Cisco 7960 over voice: the phone rings normally even hang on; then I can’t speak any with the other point and asterisk says this error:

– Got SIP response 503 “Service Unavaiable” back from
– SIP/Callmanager—43ae is circuit-busy
== Every is busy/congested at this time (1:0/1/0)

Any idea about what produces that?

The actual configuration of sip.conf is:

context=miro ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to ‘osp’
; if asterisk was compiled with OSP support.
bindport=5060 ;UDP Port to bind to (SIP standard port is 5060)
bindaddr= ; IP address to bind to ( binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
; Default is yes
;localnet=; All RFC 1918 addresses are local networks

type = friend
context = xtec
host =
port = 5061
disallow = all
allow = ulaw
allow = alaw
nat = no
canreinvite = yes
qualify = yes

and extensions.conf (I’ve tried two configurations):

exten => _4.,1,Dial(SIP/${EXTEN:1}@callmanager,60,tr)
exten => i,1,Congestion

exten => _3.,1,Macro(dialout-callmanager,${EXTEN:1})
exten => j,1,Congestion

exten => s,1,ChanIsAvail(SIP/callmanager)
exten => s,3,Dial(${AVAILCHAN}/${ARG1})
exten => s,4,Hangup
exten => s,102,Congestion

Thanks for all,