Have you tried NAT=force_rport ? Ashwin
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luca Bertoncello Sent: 07 June 2015 11:44 To: Asterisk Users Subject: [asterisk-users] Curious problem with NAT Hi list! Since the internal calls work as expected and I can register my Asterisk on an external provider, I'd like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls. Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180 Then I added the peer in my users.con: [00491771111111] fullname = 00491771111111 secret = MYVERYSECRET type=peer nat=yes qualify=yes qualifyfreq=60 hassip = yes dahdichan = 1 transport=udp,tcp callwaiting = no context = default host = dynamic dtmfmode=rfc2833 dial=SIP/00491771111111 and finally "core reload". On my Gateway I configured the NAT so: /sbin/iptables -t nat -A PREROUTING -p udp --sport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p tcp --sport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p udp --dport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p tcp --dport 6060 -j DNAT --to-destination 192.168.200.120:5060 Well, the phone connect to the server and I can see it reachable: OpenWrt*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 00491771111111/0049177111 192.168.200.3 D N 40702 OK (1768 ms) Well, now I call the mobile phone from another peer. It rings and I can answer the call. Wonderful! But no word will be sent... :( I cannot hear anything on my mobile phone and I cannot transmit a single word.... I tried to connect my mobile phone to a public VoIP-Provider and it works as expected, so I'm sure that the problem is on my network, but I can't find it... What am I doing wrong? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email (and any attachments or hyperlinks within it) may contain information that is confidential, legally privileged or otherwise protected from disclosure. If you are not the intended recipient of this email, you are not entitled to use, disclose, distribute, copy, print, disseminate or rely on this email in any way. If you have received this email in error, please notify the sender immediately by telephone or email and destroy it, and all copies of it. Thank you for your cooperation. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
