----- Original Message ----- > From: "Carlos Chavez" <cur...@telecomabmex.com> > To: asterisk-users@lists.digium.com > Sent: Thursday, August 1, 2013 8:41:19 PM > Subject: [asterisk-users] External sip phones register with the servers IP... > > We have just updated our office server to Asterisk 11.4.0 from 1.8.15 > and > internally everything is working fine. The problem we are having is > that we > cannot use any external phone connected through the Internet. This > used to > work fine with 1.8 but since the upgrade whenever you register any > phone from > an outside network the phone tries to register using the servers > internal IP. > > I endo up having something like this: > > Sending to 187.163.93.235:58545 (no NAT) > -- Registered SIP '2003' at 192.168.2.50:58545 > Reliably Transmitting (no NAT) to 192.168.2.50:58545: > OPTIONS sip:2003@192.168.2.50:58545;ob SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.50:5060;branch=z9hG4bK5f2019c0 > Max-Forwards: 70 > From: "asterisk" <sip:asterisk@192.168.2.50>;tag=as4ed13172 > To: <sip:2003@192.168.2.50:58545;ob> > Contact: <sip:asterisk@192.168.2.50:5060> > Call-ID: 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX 11.4.0 > Date: Fri, 02 Aug 2013 00:27:48 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH > Supported: replaces, timer > Content-Length: 0 > > I really cannot understand what is wrong, I have checked my sip.conf > configuration and it is the same as in past versions. externaddr and > localnet > are set to the proper values. Any ideas?
Did you look at the CHANGES file? There are new settings for NAT. If you are using the same settings as in 1.8, there is a posiblity that you will have problems depending on what settings you have (which you did not include in this message). Also, I would recommend 11.5 since there was a one-way audio issue fixed related to using the two new NAT settings. -- Michael (elguero) -- _____________________________________________________________________ -- 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
