I am new to VOIP, Linux and Asterisk. Through a lot of reading (this list, voip-info.org, documentation, etc.), I successfully installed FC3 and * on a new Dell SC420 with two X100P connecting to two PSTN lines at my office. I've also installed AMP to help me configure IVRs, call groups, extensions, etc. I use a Handytone-286 ATA and x-lite clients on the internal network and all works fine. I would like to connect to * as an extension from home, from client sites, from hotels, etc. Most of these places will be behind some type of NAT and/or firewall. At my home, for example, I have a consumer grade firewall/NAT. I cannot get the Handytone-286 to work properly from there. I connect to the * server and register, I can call out and incoming calls ring in, but there is no audio sent nor received regardless of whether dialing out or calling in. I suspect this has to do with RTP and how my home firewall/NAT handles RTP. Is my thinking correct here? What's frustrating is that I can't get it to work even if I put the Handytone-286 in a DMZ. Maybe the firewall/NAT is still processing and malforming the RTP packets? Even if I do get the ATA working fine behind my home NAT, I would have to do some reconfiguration most likely anywhere else I try plugging it in, right? And, if I wanted to add another ATA at home connected to the same remote * server, it's most like not going to work without custom RTP port forwards, etc., right? Thanks, John John Huang
> Even if I do get the ATA working fine behind my home NAT, I would have > to do some reconfiguration most likely anywhere else I try plugging it > in, right? And, if I wanted to add another ATA at home connected to the > same remote * server, it's most like not going to work without custom > RTP port forwards, etc., right?Yes, this is the nasty secret of SIP. Well not that secret since hundreds of pages have been written about it and several people come to #asterisk daily with this exact complaint. One direction is IAX2 and hopefully there will be the Farfon adapter in the not too distant future for that as well as the first IAX2 phones about to ship RealSoonNow. For only home use, you should be able to get it working with port forwarding.
Helder Rogério [MICROREDE]
2005-Jan-12 03:51 UTC
[Asterisk-Users] SIP, * and clients behind NAT
Hi John, I had a similar problem solved while putting on the extension of the terminal adapter nat=yes [252309970] type=friend host=dynamic callerid="252309970 - Pincol Sede" <252309970> nat=yes canreinvite=yes Hope it solves your prob ----- Original Message ----- From: "John Huang" <asterisk@thinkapex.com> To: <asterisk-users@lists.digium.com> Sent: Tuesday, January 11, 2005 5:33 PM Subject: [Asterisk-Users] SIP, * and clients behind NAT> I am new to VOIP, Linux and Asterisk. Through a lot of reading (this > list, voip-info.org, documentation, etc.), I successfully installed FC3 > and * on a new Dell SC420 with two X100P connecting to two PSTN lines at > my office. I've also installed AMP to help me configure IVRs, call > groups, extensions, etc. > > I use a Handytone-286 ATA and x-lite clients on the internal network and > all works fine. > > I would like to connect to * as an extension from home, from client > sites, from hotels, etc. Most of these places will be behind some type > of NAT and/or firewall. At my home, for example, I have a consumer > grade firewall/NAT. I cannot get the Handytone-286 to work properly > from there. I connect to the * server and register, I can call out and > incoming calls ring in, but there is no audio sent nor received > regardless of whether dialing out or calling in. > > I suspect this has to do with RTP and how my home firewall/NAT handles > RTP. Is my thinking correct here? What's frustrating is that I can't > get it to work even if I put the Handytone-286 in a DMZ. Maybe the > firewall/NAT is still processing and malforming the RTP packets? > > Even if I do get the ATA working fine behind my home NAT, I would have > to do some reconfiguration most likely anywhere else I try plugging it > in, right? And, if I wanted to add another ATA at home connected to the > same remote * server, it's most like not going to work without custom > RTP port forwards, etc., right? > > Thanks, > > John > > John Huang > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >