Hi all
I am in this strange situation: we had ser configured to relay calls to
numbers to asterisk extensions and all used to work nicely, with both ser and
asterisk running on the same machine with public ip (ser on port 5060 and *
on 5061). We had to move temporarily our server to another provider which put
our server on a dmz, so that now we have our server with private ip but
reachable from the outside via port forwarding on a public ip. Now every
communication with asterisk is mute, calls are relayed by ser, connections
estabilished, but no voice either with sip or with demo-echotest (* log says
he is playing echotest but I can't hear anything!). I thought this was a dmz
firewall + rtp problem but ports in rtp.conf are open with forwarding (udp).
This is current network situation:
myserver: private ip 10.0.0.229, ser running on port 5060, asterisk on 5061
(sip), rtp ports 5082-5092.
Public ip 82.184.xx.xx with udp forwarding on above ports to myserver private
ip
Ser listens to private ip address and forwards to asterisk on private ip
(can't forward to public address). bindaddress in sip.conf =private ip or
0.0.0.0 (mute in both cases), can't bind on public ip.
Intresting part in sip.conf
[general]
port = 5061 ; Port to bind to
;bindaddr = 10.0.0.229 ; Address to bind SIP channel to
bindaddr = 0.0.0.0
context = 82.184.xx.xx ; Default context for incoming calls
srvlookup = no ; Enable DNS SRV lookups on outbound calls
;;;;;;; tried with or without following lines, still mute :-(
autocreatepeer=yes
externip=82.184.xx.xx
register => asterisk:xxxxx@10.0.0.229/100 ;asterisk actually registers on
ser! realm=82.184.xx.xx
;;;;;;; tried also with public ip host, nat=no, canreinvite=yes, type=peer
[asterisk]
type=friend
secret=xxxxx
username=asterisk
host=10.0.0.229
nat=yes
canreinvite=no
;dtmfmode=rfc2833
I hope someone can give me a hint to resolve this crappy situation
thanks a lot
--
Giovanni Balasso
giaso apud supereva.it
