Eric,
The problem you are seeing is because the RTP (voice) packets being sent
towards the NAT'd UA are being blocked by the NAT router.
The UA being used behind NAT will need to have a static IP address set
(e.g. 192.168.1.50) and on the NAT router you will need to permanently
forward (port forward) SIP and RTP ports to the internal IP address. I
recommend ports 5060 and 16384-16400. On the NAT'd UA, set the SIP port
to 5060 and the RTP ports to 16384-16400. If your UA only supports one
RTP port, just use 16384.
As Forrest noted, you will also want to set canreinvite=no in sip.conf
for the NAT'd UA. You should also set nat=yes, which will force asterisk
to re-write SIP packets coming from the NAT'd UA to the correct external
IP address of the NAT router.
-mike
Eric Yu-Wei Sung wrote:
I am trying to make 1 soft SIP UA behind NAT connect to a public hard
CISCO UA via a public asterisk server. The CISCO UA can hear the voice
from the SIP UA but not vice versa. I do set nat to yes for the soft
phone. Any help would be greatly appreciated.
Below is my sip.conf
[general]
port = 8060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[2000] ; soft phone behind NAT
type=friend ; This device takes and makes calls
username=2000 ; Username on device
host=dynamic ; This host is not on the same IP addr every time
context=from-sip ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
nat=yes
[2002] ; CISCO hard phone
type=friend
username=2002
secret=2002
host=dynamic
context=from-sip
mailbox=103
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