similar to: VPN Pass through documentation

Hi everyone. I am so baffled by the following problem: Office 1 is using ADSL and it is building a VPN tunnel with IPSEC to Office 2. Both ends are using shorewall/freeswan firewalls. Diagram: Office1 fw --- VPN TUNNEL --- Office2 fw --- cisco router ----- VLANS | DMZ Office 1 has the following interfaces: 2: eth0:

Hi, This subject has been brought up in the forum, but it''s a bit different. If I have a set of tun interfaces. I already defined tun+ as zone A, and I have excluded tun15 as zone B (a subset of zone A). I need to add tun16 to zone B. My config: /etc/shorewall/interfaces: A tun+ - routeback B tun15 /etc/shorewall/ A ipv4 B:A ipv4 I tried to define in

Hi, I have a rule: DNAT net dmz:a.b.c.d tcp 25 - k.l.m.n The problem: I want to DNAT port 25, 143, 110 k.l.m.n is alternate public ip (using vrrp, just like alias) Can I abridge the above line using macro, instead of writing 3 separate lines? I can do: MailPorts/DNAT net dmz:a.b.c.d But this refer to the physical public ip I have more elaborate requirement to forward around 20

Hi. I am trying to force some traffic that goes to address 203.7.93.94 through a VPN tunnel. I use freeswan 1.98b and Shorewall 1.4.6c in one machine. The 203.7.93.94 is in the DMZ on the other end. (Both ends use the same shorewall and freeswan). I have successfully set up a tunnel between the two network (using a point to point topology, not hub). I added a static routing that redirect

Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Thanks. Lito -------------- next part

If you are having trouble reading this email, read the online version<http://now.eloqua.com/es.asp?s=491&e=78675&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> . <http://app.en25.com/e/er.aspx?s=491&lid=215&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> Dear Lito, *The information in this email is given to you in advance to make you aware of an impending product release

2017-04-21 1:08 GMT-03:00 Epsilon Minus <theepsilonminus at gmail.com>: > 2017-04-21 0:38 GMT-03:00 Epsilon Minus <theepsilonminus at gmail.com>: >> >> >> I'm desperate. My English is terrible, but I will try to be as clear as possible. >> >> I have Samba 4.4.4 in a Centos 7. >> >> I have many connections for each ip. >> >>

I'm desperate. My English is terrible, but I will try to be as clear as possible. I have Samba 4.4.4 in a Centos 7. I have many connections for each ip. The command smbstatus -b returns me infinity of these: Samba version 4.4.4 PID Username Group Machine Protocol Version Encryption Signing

This might be a newbie question but I'm just wondering how would it be possible to have 30 analog lines using asterisk for PBX by just using TDM40B and X100P (or are there any device>), if an ordinary PC support just 4 PCI slots? the maximum scale i guess would just be 2 x 8. Adding a new PC just for this purpose would be costly. I would appreciate your comments. Thanks.

> Lito Lampitoc wrote: > > thanks for enlightening. So you mean, if I have 3 lines when the > caller > > dialled the first line and it was busy, the call will be diverted > to the > > next two available lines in random? > > > > I don't think it's random. I think its just sequential. If main > line > is busy, try second. If that is

After some investigation I found out that constrained ordination is working fine under R version 6.5.0 and 6.5.1 but not under any of newer versions. Is it posible to repair/make something in for example newest version of R 7.0 lito enble correct using of constrained ordination under BiodiversityRGUI environment? [[alternative HTML version deleted]]

2017-04-21 0:38 GMT-03:00 Epsilon Minus <theepsilonminus at gmail.com>: > > > I'm desperate. My English is terrible, but I will try to be as clear as possible. > > I have Samba 4.4.4 in a Centos 7. > > I have many connections for each ip. > > The command smbstatus -b returns me infinity of these: > > > Samba version 4.4.4 > PID Username

I seem to have found the solution. But I have not the slightest idea what I have to do Does anyone help me? https://bugzilla.samba.org/show_bug.cgi?id=11472 Thanks! 2017-04-21 1:16 GMT-03:00 Epsilon Minus <theepsilonminus at gmail.com>: > 2017-04-21 1:08 GMT-03:00 Epsilon Minus <theepsilonminus at gmail.com>: >> 2017-04-21 0:38 GMT-03:00 Epsilon Minus <theepsilonminus

Hello @r-devel, ??? From what I read in the following thread: https://stat.ethz.ch/pipermail/r-package-devel/2023q4/010197.html ??? The major reason to limit paths lengths in R packages is to help support MS Windows 260-chars paths lengths limit (pre-W10). ??? However, the official documentation at this address: https://cran.r-project.org/doc/manuals/R-exts.html#Package-structure-1 ???

hi all, I successfully deployed the ROR project on Centos server,i have one question how can i access that application from my machine ? on starting WEBRick server it is started on this ip http://0.0.0.0:3006, i want to set my machine ip in place of ''http://0.0.0.0:3006'' and also how i can access it from my machine ? Thanks in advanced. Thanks and Regards Sachin S.

When I upload a pre recorded wav file using trixbox, it can't be played on the welcome message. But when I record using xlite, it works ok. trixbox required 8Khz PCM 16bit recording, I used it, but still no success. Any idea? Thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there, Im getting this notice in CLI, but the call quality is okey, Im using digium TE406 and asterisk 1.2.4. here are the CLI actual logs: -- Executing SetAccount("Local/50015308467418@default-ca2e,2", "XXXXXX") in new stack -- Executing AGI("Local/50015308467418@default-ca2e,2", "call_log.agi|50015308467418") in new stack -- Launched AGI

Has anyone here successfully tried this? hipath 3750 --> hg1500 --> asterisk i'm not sure with the flowlines though. Thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060629/11f85674/attachment.htm