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 213.176.191.21
– 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:
[general]
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=0.0.0.0 ; IP address to bind to (0.0.0.0 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
auth=md5
realm=miro.cat
;localnet=213.176.0.0/255.255.0.0; All RFC 1918 addresses are local networks
…
[callmanager]
type = friend
context = xtec
host = 213.176.191.21
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
[macro-dialout-callmanager]
exten => s,1,ChanIsAvail(SIP/callmanager)
exten => s,2,CUT(AVAILCHAN=AVAILCHAN,1)
exten => s,3,Dial(${AVAILCHAN}/${ARG1})
exten => s,4,Hangup
exten => s,102,Congestion
…
Thanks for all,
Pep