search for: natted

Hi i have a debian box working as a router.. it works quite well, now i want to give more than 1 ip.. is it possible to do it? some of them must be an open ip.. i mean.. all ports opened is it possible? how should i do it? Here is my nat.sh script just in case someone wants it.. (comments r in spanish.. and not right) Thanks in advance, #!/bin/sh echo "AthoS LaN Generando

Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? Thanks..

I've been lurking on the dev discussion on creating nat=auto. It all leads me to think there's no reason to use nat=no. We have about 60 internal sip extensions connected to an multihomed asterisk box where the external ip is not nat'ed. Each of the internal sip contexts has nat=no. On startup I get a slew of warnings about intruders being able to distinguish real extensions. But

Asterisk Project Security Advisory - AST-2011-013 Product Asterisk Summary Possible remote enumeration of SIP endpoints with differing NAT settings Nature of Advisory Unauthorized data disclosure Susceptibility Remote

https://bugzilla.netfilter.org/show_bug.cgi?id=1213 Bug ID: 1213 Summary: Nft stateless NAT (NOTRACK) Product: nftables Version: unspecified Hardware: All OS: Ubuntu Status: NEW Severity: critical Priority: P5 Component: nft Assignee: pablo at netfilter.org Reporter:

https://bugzilla.netfilter.org/show_bug.cgi?id=821 Summary: Rosa Product: iptables Version: unspecified Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P5 Component: iptables AssignedTo: netfilter-buglog at lists.netfilter.org ReportedBy: andoandre at gmail.com

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??

...Does it use one port for both commands and voice packets? Does the remote server just use the "received from" IP address and port to respond? Finally, would an IAXy work seamlessly in a configuration where it is plugged into a NAT router which is plugged into another NAT router - double NATted? The * server is on a public IP. -- Nabeel Jafferali tel: 647.722.8457 x201 718.606.4190 x201 fwd: 46990 x201 email/msn: nabeel<at>jafferali.net

I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 10000 - 30000 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see

I'm at my wit's end! I've spent 2 days now trying to get what I thought was a very simply SIP + NAT arrangement working. I've trawled the web and picked brains, but nothing anyone suggests work. My setup is very simple. I have a * server in a datacentre, with a public IP address. There is no firewall in place, it's completely open (at least, as far as I'm concerned). I

Hi to everybody my system is be attack, but I dont know what this means [May 27 15:12:24] WARNING[26018] chan_skinny.c: Partial data received, waiting (76 bytes read of 786) [chan_skinny.c] skinny_session[0][C-00000000] skinny_session: WARNING[May 27 15:52:32] Asterisk 13.8.0 built by root @ asterisk on a x86_64 running Linux on 2016-04-04 19:02:51 UTC [May 27 15:52:32] NOTICE[2306] cdr.c: CDR

This week, I've been really busy with the launch of a new Swedish Voip provider, www.bbtele.se, so I haven't been able to follow the Asterisk community and haven't been very responsive either. My apologies if you've tried to contact me and I did not reply quickly or at all. So to cover up (can't report on what is happening :-) I dedicate this issue of Asterisk Sunday News to

Hi All; I succeeded to have a success call from Polycom behind NAT while Asterisk has public IP address, but I was not able to have a succeed call (it was established, but no voice running, and then the call disconnected) if Asterisk behind NAT and Polycom behind NAT. When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to asterisk (at asterisk router) and to Polycom IP

Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. -------------- next part -------------- An HTML attachment was

Hi Ray, I was wondering if the "qualify" option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being

I know that NAT is something no one really likes to talk about, but does anyone know how work with it elegantly? There are many providers which deal with it on a daily basis in fact they cater to it, is this possible to do with asterisk or does it require other exotic setups? I even know of a provider which uses asterisk with many different types of devices, and they handle all NAT config on

Hello, Are there any good alternatives to the network-nat script on Debian Squeeze? I have tried using (network-script network-nat) and (vif-script vif-nat), but I end up with an error message like Error: Device 1 (vif) could not be connected. ip addr add 10.0.1.259 dev vif4.1 failed with configuration file directive vif =[ ''script=vif-nat,ip=10.0.1.132'' ] I need Internet

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

Hi, everyone, We've been reasonably happy with Polycom SoundPoint phones, but we only have them installed on the LAN. I've read that they have problems working across NAT. So ... I guess I have a few questions. First, is there a way to get Polycoms to work well over NAT? If not, then are there phones of comparable voice quality that do work well over NAT? Without costing a lot more?