similar to: looking for a tool to tunnel ethernet reliable on internet.

http://bugzilla.mindrot.org/show_bug.cgi?id=1231 Summary: "Tunnel ethernet" always uses next tap-device Product: Portable OpenSSH Version: 4.3p2 Platform: ix86 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: bitbucket at mindrot.org

http://bugzilla.mindrot.org/show_bug.cgi?id=1231 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |ASSIGNED Keywords| |help-wanted CC|

https://bugzilla.mindrot.org/show_bug.cgi?id=1231 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|ASSIGNED |RESOLVED Resolution| |WORKSFORME --- Comment #2 from Damien Miller <djm

Hi (I'm not subscribed, please, cc) I've got an ethernet level ssh tunnel to a remote system. That system is running a terminal server, so, that tunnel is the only connection, I've got. The terminal server is connected via a serial and an ethernet cable to a test board and is configured to bridge that ethernet connection to my tunnel. I can then trigger that test board to start a

I used the "FreePBX on Debian" HowTo at http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles to initiate calls to my SIP carrier. They get my registration, but they see that my call is interrupted before they can complete the connection. My Asterisk log shows that the call times out after the time (45s) specified in my dialplan Dial() command. What is wrong? [from

Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '7ad7ce9a09aebcce6e1c3e551fa4d401@192.168.1.1' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '06d1853a435e01fb05c3233d7aecba85@192.168.1.1' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on

Hello list, I need to provide internet access through a proxy server on a central office to a remote Lan on a branch office (LAN-B). Also there is an internal server that LAN-B machines should reach. Below there is a simple diagram. Right now I have an IPsec VPN tunnel between offices, and LAN-B can acces LAN-A machines without problems. But LAN-B machines can't access the remote proxy or

Hi, I have searched the archived, and see this question has come up with some frequency, but the solution by the original poster is never posted or the question is not answered completely. I have also RTFM (rsync and ssh), and I've been trying examples that I have googled. So here it is again. BoxA --internet-- BoxB (accessable only by ssh) -- BoxC I want to backup BoxC to BoxA. I want

Hello, I have a laptop with two network interfaces: eth0 (LAN) and eth1 (WAVELAN). Now I am looking for a way to route ssh and icpm packages via the wavelan interface and the rest over the LAN interface. My problem is now that every applications binds eth0 interface as default (which results in source addr 192.168.0.2). What I did is using SNAT to change source-address to 192.168.1.2 for packages

On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4h07 at gmail.com> wrote: > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>: > >

Hi Wayne, I'm not subscribed anymore, but I once sent you this example/FAQ. I searched the archives (because I needed to implement this) and noticed I never posted the solution (like a dummy). Would you post it for me? Thanks Jay Backing up through a firewall Its very common to have a setup where you can not reach the machine you want to backup directly, because it is behind a

>> The call that gets dropped had a retransmission of INVITE from UAC >> to UAS (and therefore retransmission of 200 OK from UAS to UAC). >> There is nothing wrong with the re-transmission as such, but I >> noticed a potential bug in Asterisk in the way it responds to an >> INVITE retransmission. Asterisk is bumping up the session version >> number in

I'm getting a couple strange things with asterisk on Broadvoice. It works fine currently for inbound and outbound. Everything is registered, its perfect. My problems currently are: 1) After 20 minutes audio out to broadvoice goes away 2) After about 3 hours my registration attempts to broadvoice time out and the only solution is to switch to a different proxy which subsequently works.

root at ne-vlezay80:~# tcpdump -i tap0 -n tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on tap0, link-type EN10MB (Ethernet), capture size 262144 bytes 00:23:53.206672 ARP, Request who-has 10.194.0.2 tell 10.194.0.200, length 28 00:23:53.206691 ARP, Reply 10.194.0.2 is-at 52:54:00:38:b9:0b, length 28 00:23:53.710691 STP 802.1d, Config, Flags [none], bridge-id

I am hoping some of you can help me troubleshoot this problem I am having with my home asterisk machine. I have incoming POTS service using a SPA-3000 (extension 119). Calls on that line go to an attendant recording that offers a menu choice: press 1 for Nancy, press 2 for the rest of us. In reality, pressing anything other than 1 sends the call to the rest of us by dialing both extensions 101

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

https://bugzilla.mindrot.org/show_bug.cgi?id=2365 Bug ID: 2365 Summary: openssh client ignores -o Tunnel=ethernet option, creating an IP tunnel device instead of an ethernet tap device Product: Portable OpenSSH Version: 6.6p1 Hardware: amd64 OS: Linux Status: NEW

Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '765e20595817e9897b77cff23f821cc5@10.0.0.254' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '4858cde16223cc0716e325921a8a0654@10.0.0.254' of Request

I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found

can you please tell me why this is not working? i got the example for how to use each online and copied it to a t (except i added the if). its telling me items.each does not exist. var items = document.getElementsByClassName(''editView''); if (items.length > 0){ items.each(function (result){ thisid = result.id; thiskey = thisid.replace(/se_item_/gi, '''');