[asterisk-users] Sip phones on localnet AND outside localnet problem
Hi list I am having trouble getting asterisk to perceive the firewall's ip address as outside localnet (setting in sip.conf). The situation is this: - phones inside lan work fine when localnet is set to 192.168.0.0/255.255.255.0 - phones outside the lan can't ack the invite from asterisk because asterisk perceives the outside phone to originate from localnet as DNAT is done with iptables and the incoming ip address is rewritten (i'm assuiming) to the firewall's internal ip address (192.168.0.20) (so from and contact are 192.168.0.21, this is the asterisk box's private ip) - If i set localnet to 10.0.0.0/255 asterisk puts the externip in the from and contact fields and it works fine with an external phone - I haven't tested it yet, but i don't think a sip phone in the lan with the asterisk box will work if an incorrect localnet is set! Can anyone help or offer a workaround for this to allow sip phones in the lan as well as outside to work? Is it possible to mangle the packets with iptables in postrouting so that they don't have the firewall's ip address when received by asterisk? Is there a setting in sip.conf that can make the firewall ip appear as non-local even though it is in the localnet setting? Thanks for any help! Marcus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk keeps sending invite to sip phone "No response to critical packet"
Thanks Alex I suspected that no ack was being sent/received too. The invites are getting sent to the phone but nothing is coming back from the phone to the firewall. Does anybody know how I can sniff packets being sent and received to/from the phone and/or modem router the phone is connected to? I can't upload software to the modem/router or the phone (ie a packet sniffer). Can packets be sniffed from a linux box on the lan (please excuse my ignorance!)? If so can anybody point me to a resource that may help? Thanks for any help Marcus Asterisk is not receiving replies to the INVITE - probably due to NAT issues. marcus wells wrote: > Hi there > > I am wondering if anybody can help me illuminate a problem I am having > with my asterisk installation. I am using: > > - IP phone (Siemens gigaset S685IP) behind a modem/router that has ports > udp 5060 and 1:10100 forwarded to the static ip of the IP phone > (192.168.0.3). This has to go to: > - modem that operates in half bridge mode (no nat) to a linux firewall > (does natting ip is 192.168.0.20) that has the ports above forwarded to > the staitc ip of the asterisk box (192.168.0.21 packaged version for > ubuntu hardy). > > This phone works fine with a commercial provider of viop (via asterisk), > but I can't get it to work with my install of asterisk in my remote network! > > ngrep-ing traffic on the firewall shows asterisk continually sending > invites to the public ip of the ip phone. > > I would be very grateful for any pointers of where to start. If you need > sip debug or ngrep info let me know and i'll reply with it. I've been > beating my head against this for some time now! > > Thanks in advance > > Marcus > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk keeps sending invite to sip phone "No response to critical packet"
Hi there I am wondering if anybody can help me illuminate a problem I am having with my asterisk installation. I am using: - IP phone (Siemens gigaset S685IP) behind a modem/router that has ports udp 5060 and 1:10100 forwarded to the static ip of the IP phone (192.168.0.3). This has to go to: - modem that operates in half bridge mode (no nat) to a linux firewall (does natting ip is 192.168.0.20) that has the ports above forwarded to the staitc ip of the asterisk box (192.168.0.21 packaged version for ubuntu hardy). This phone works fine with a commercial provider of viop (via asterisk), but I can't get it to work with my install of asterisk in my remote network! ngrep-ing traffic on the firewall shows asterisk continually sending invites to the public ip of the ip phone. I would be very grateful for any pointers of where to start. If you need sip debug or ngrep info let me know and i'll reply with it. I've been beating my head against this for some time now! Thanks in advance Marcus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
