similar to: Problem with NAT - Part 2

Have you tried NAT=force_rport ? Ashwin -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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

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

Hi list! Today I tried to change the NAT-configuration on my Firewall to use another port for SIP. I configured it so: /sbin/iptables -t nat -A PREROUTING -p udp -m udp --dport 10000:10100 -j DNAT --to-destination 192.168.20.120 /sbin/iptables -t nat -A PREROUTING -p udp -m udp --dport <my new port> -j DNAT --to-destination 192.168.20.120:5060 then, I tried to log on my Asterisk

Hi again, list! I know, I'm really annoying the list... :) Well, maybe I got my Asterisk at home ("wrt" on the previous E-Mails) to accept my mobile phone from Internet. It was a problem with the network and the firewall. Now I can log my mobile phone in my Asterisk in and the phone is REACHABLE. Wow! Got it! If I call a phone at home using my cellphone it works and the

> Am 14.06.2020 um 16:38 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 22:56 schrieb Antony Stone: > > Hi again, > >> 2b. Take your Thomson telephone to some other location with Internet access, >> let it register to your home Asterisk server, and them make a call to the same >> number yet again. I'm sure you can get

For such cases i created a dialplan in the default dialplan which blocks the ip of the hacker with iptables. On Monday, June 8, 2015, Luca Bertoncello <lucabert at lucabert.de> wrote: > Hi list! > > Very strange... > I ran the Asterisk CLI for other tasks, and suddenly I got this message: > > == Using SIP RTP CoS mark 5 > -- Executing [000972592603325 at

I'm guessing this is a small/home system? I suggest you install SecAst from this site: www.telium.ca It's free for small office / home office and will deal with these types of attacks and more. It can also block users based on their Geographic location (based on the phone number it attempted to dial I suspect this is middle east), look for suspicious dialing patterns, etc. If you

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=498 Summary: RTP packets are not hitting NAT table Product: netfilter/iptables Version: linux-2.6.x Platform: All OS/Version: Fedora Status: NEW Severity: major Priority: P2 Component: NAT AssignedTo: laforge@netfilter.org ReportedBy:

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi Antony, > I would like to see a much simpler one-for-one comparison: only change one > thing at a time, and see what the difference is. > > So: I suggest you try *two* independent *pairs* of tests: OK > 1a. Using your Android phone, connect using your home wireless network (I > assume you have a wireless network, if not then

https://bugzilla.netfilter.org/show_bug.cgi?id=1117 Bug ID: 1117 Summary: Table ipv4-nat prerouting dnat doesn't accept dest IP:PORT Product: nftables Version: unspecified Hardware: x86_64 OS: All Status: NEW Severity: enhancement Priority: P5 Component: nft

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the same > number yet again. I'm sure you can get the Thomson to connect to Asterisk via > some external network, since you say you can do this from your Android phone.

Am 11.06.2019 um 20:42 schrieb Antony Stone: Hi Antony, > I think the main question here is: how are you connecting Asterisk to the > telephone system? Via VoIP... > You mention that you're on DSL from Deutsche Telekom, but is the call going > over this DSL link to soem SIP provider, who then connects you to the PSTN, or > are you connecting Asterisk locally to the phone

Am 11.06.2019 um 21:10 schrieb Antony Stone: Hi, > So, you have a SIP phone, connected to an Asterisk server on your local > network, which then connects to D Telekom's SIP server over the DSL line? Correct! >> The other party use VoIP, too, since they are in Germany (and Italy) and >> here there are just VoIP... Sigh! > > Are they also using a SIP phone? My

Hi list! Am 22.06.2020 um 16:48 schrieb Luca Bertoncello: > Hi list! > > So, now I have a business contract and a technician was here to check > the DSL... > Nothing found, except that for 50Mbps I need now vectoring. Really > nice... A couple of years ago I could get 50Mbps without vectoring. > Of course, Deutsche Telekom said nothing about this change... > > Well, I

What settings have you got for directmedia? Could you try nat=force_rport,comedia directmedia=no -Ashwin -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Luca Bertoncello Sent: 07 June 2015 12:04 To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Curious problem with NAT

On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > > > What settings have you got for directmedia? > > > > Could you try > > > > nat=force_rport,comedia > > directmedia=no > > Tried. Peer always unreachable, call not possible... :( > >

On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > > Are you using the wifi on on the cellphone? The peer IP is showing as >> 192.168.200.3 which is not a routable address. Unless things have >> changed, >> double NAT configurations do not work. >> >

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

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

Hi list! Very strange... I ran the Asterisk CLI for other tasks, and suddenly I got this message: == Using SIP RTP CoS mark 5 -- Executing [000972592603325 at default:1] Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") in new stack == PROXY Call from 0123456 to 000972592603325 -- Executing [000972592603325 at default:2]