[Asterisk-Users] SIP NAT question

[George Lin]

Hello all,

I am sorry to bring the old question to the community. But I cannot find any
answer in the google.

I want to deploy multiple SIPs phone in our office. And we have shutdown the
firewall at our office router(with ip 211.x.x.x). we have deployed the
asterisk with IP 218.x.x.x.

All SIP phones have 192.x.x.x.

When the SIP phone is power on, they are registered in the asterisk. we can
check at asterisk side by issueing "sip show peers", and all the phones are
associated with 211.x.x.x:port-number.

PRoblem:
Now some times the sip can receive call, and some time it cannot recieve
call. When we dumping the sip log, and see that asterisk tried to INVITE the
specified SIP phone with the 211.x.x.x:port-number, and was failed after 5
times. But the call orginated from SIP phone is always OK.

Questions are:

1. Does asterisk remember the mapping between 192.x.x.x AND
211.x.x.x:port-number ?
2. When a call to a sip phone, is it asterisk responsiblility to map the
211.x.x.x:port-number to the 192.x.x.x, and send to the office router ? OR
it is the office router to remeber all the mapping between each sip phone
192.x.x.x and 211.x.x.x:port-number, and asterisk juts sends the
211.x.x.x:port-number to the office router ??
3. If it is the office router's responsiblity, what should we configure the
office router even there is no firewall???

Please advise , and thanks alot.

George Lin