Hi, ultimately I want to set up a Asterisk system with 9 analog phones connected into a ATA which connects to the SIP provider via asterisk. This is so I can record all outgoing calls.
The ATA I am using is a SPA-8000 and my SIP provider is telecube. So far I have got one analog phone working with one extension and am trying to set up the second. When I comment out the first to test the second I get congestion. When i revert back to the first configuration i sometimes get congestion but occasionally it works. I do not know what I am doing wrong. I have 9 SIP extensions with telecube but am testing 2 just to see if that works for now.
Can someone please look at what I’ve done and help me out?
SIP.conf
[code][general]
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no
;register = :@telecube-extension-0
register = :@telecube-extension-1
office-phone
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfMode=auto
disallow=all
allow=g722
allow=ulaw
allow=alaw
101
secret=101
102
secret=102
;–
[telecube-extension-0]
type = peer
host = sip.telecube.net.au
insecure = invite
dtmfmode = rfc2833
disallow = all
allow = ulaw
defaultuser = ***
secret = ***
–;
[telecube-extension-1]
type = peer
host = sip.telecube.net.au
insecure = invite
dtmfmode = rfc2833
disallow = all
allow = ulaw
defaultuser = ***
secret = ***[/code]
Extensions.conf
[code][LocalSets]
exten => 101,1,Dial(SIP/101)
exten => 102,1,Dial(SIP/102)
exten => 200,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
exten => _04XXXXXXXX,1,MixMonitor(/home/${EXTEN}.wav,b,a)
;same = n,Dial(SIP/${EXTEN}@telecube-extension-0)
same = n,Dial(SIP/${EXTEN}@telecube-extension-1)[/code]