Forgot to mention that we have specified the nat=yes for all sip entries in sip.conf.
Regards -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of George Lin Sent: Wednesday, August 13, 2003 10:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP NAT question 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
