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Dear All Why NAT rules stays valid even if I flush nat anf table chains?? I have: iptables -P FORWARD DROP iptables -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A FORWARD -s SOME_IP -d SOME_BCP_5_IP --dport 1234 -j ACCEPT iptables -i nat -A PREROUTING -s SOME_IP -d MY_INTERNET_IP \\ --dport 1234 -j DNAT --to-destination SOME_BCP_5_IP The conection is

On Mon, 18 Jan 2016, Ethy H. Brito wrote: >> how to flush user input before READ()? How about a read() to a dummy variable with a 1 second timeout to consume the octothorpe and password? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi All Can someone tell me what am I doing wrong? These was suposed to catch icmp type 8. Why ''match u8'' does not work? # tc filter add dev eth0 protocol ip parent 1: prio 2 \ u32 \ match u16 0x0800 0xffff at -2 \ match u8 8 0xff at 20 \ flowid 1:5 Illegal "match" # # tc filter add dev eth0 protocol ip parent 1: prio 2 \ u32 \ match

Well, things seem to be going really well with tbf, prio and sfq. But I''m a nosey bugger. :-) I''d love to be able to audit what''s going through each of the prio bands. The super ideal solution would be to be able to attach tcpdump to each band and see what''s going through it with the benefit of tcpdump''s filtering so that I can examine and filter

snat not working my local ip is aaa.aaa.aaa.aaa asterisk sitting on the internet at ip bbb.bbb.bbb.bbb my firewall''s internal ip is 192.168.0.254 i did snat: iptables -t nat -A POSTROUTING -o ppp0 -j SNAT --to aaa.aaa.aaa iptables -t nat -L -v gives: Chain POSTROUTING (policy ACCEPT 23663 packets, 2182K bytes) pkts bytes target prot opt in out source destination 33056

Hi I had a bad experience upgrading Ubuntu a few months ago. Today I made a "dd" copy to another harddisk and tried to dist-upgrade. I get "Illegal instruction (core dumped)" running "service asterisk debug" at random places. Any help will be appreciated to spot this problem. I think it is worth to mention that the dadhi hardware is not present at the copied

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there

I am running Gentoo. I am trying to get my /etc/fstab to automatically mount a remote Samba share at boot time. I have the following in my /etc/fstab file: //192.168.0.2/share /mnt/share cifs auto,credentials=/home/user/.smbpasswd,uid=1000,umask=002,user 0 0 This works fine if I specify the username and password in the /etc/fstab file. However, if I try to use the credentials file

Hi All, I wonder can I do NAT+mangle+tc on same maschine? I want to shape outgoing traffic per IP on my gateway computer. Regards Emil Terziev _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Hi. we are experimenting a strange issue in our PBX. By example: if we dial to the 100, the call is answered in 199. We dont have any redirection for that, but the cli show the same issue when request show peers. Aditionally, the user 100 use the ip address 192.168.11.100, and the cli show connected the user from 192.168.11.160 (that ip is assigned to the user 199) PBX*CLI> sip show peers

Hi how to flush user input before READ()? I wrote a small script to ask for user password before granting access to outside, but some telefones memorize the full user input, including "#". So, when the user press redial, for instance 5556789#123, asterisk accepts the number and the password "123" and gives access to the outside word to whomever redials that terminal. Any

On Thu, 28 May 2015 11:15:45 -0500 Scott Griepentrog <sgriepentrog at digium.com> wrote: > The string "5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060" is the unique > identifier for the call in SIP known as the Call-ID. If you have a packet > capture of the port 5060 SIP traffic, that identifier will be in each SIP > message related to the call, which also

Hi All I've installed Samba 2.0.3 and coudn't put "unix passowrd sync" to work. I set: passwd program = /usr/bin/passwd %u passwd chat = *New*password:* %n\n *Re-enter*new*password:* %n\n*changed.* passwd chat debug = Yes unix password sync = Yes log level = 100 The password page says "The password for 'user' has been changed". In fact it has been

Hi How does Windows interpret the 'x' attribute for the file's 'other' attribute field? If I give a *file* a 0775 permission Windows XP clients cannot see the file. Not even the group members. If I change this permission to 0774 the file becomes visible to everyone. (smbclient sees the file either way). This file in under a directory which has 2775 as permission and

Hi All I have a few lines like this at asterisk/messages. [May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call 5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Since we have hundreds of clients with hundreds of simultaneous calls, how is it possible to know to which customer/IP

On Thu, 22 Oct 2015 15:06:59 +0300 Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Wed, Oct 21, 2015 at 01:33:27PM -0200, Ethy H. Brito wrote: > > > > Hi > > > > Who should insert dahdi_echocan_oslec.ko module in Ubuntu 14.04? > > dahdi_echocan_oslec should be built by default. What you may miss is the > 'echo.ko' (OSLEC) kernel module.

Hi all I am using a pass trhu router and I need to QoS some clients output by its IP address. The problem is that QoS is due after NATing. Is there some clever way of doing this besides MARKing every packet with some IP hashing in POSTROUTING NAT table? Regards Ethy

Hello. Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve. Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11. In asterisk 13 I did not find a handler after the call, but before

I visited yesican.chsoft.biz and the author proposes a way to match packets by less than some size . Here is the thing: match u16 0x0000 0xffb0 at 2 With this match he says that packet with less than 80 bytes will match the rule. Well, 0xffb0 translates to 1111 1111 1011 0000 (which is -80 BTW). So, if I am correct any packet with bits 4 and/or 5 set (together with any of the 4

On Mon, 18 Jan 2016 16:09:17 -0200 "Ethy H. Brito" <ethy.brito at inexo.com.br> wrote: > On Mon, 18 Jan 2016 09:38:52 -0800 (PST) > Steve Edwards <asterisk.org at sedwards.com> wrote: > > > On Mon, 18 Jan 2016, Ethy H. Brito wrote: > > > > >> how to flush user input before READ()? > > > > How about a read() to a dummy variable