similar to: Pre-PXE loader

I'm using the dovecot available on Ubuntu Feisty (1.0rc17) and am trying to migrate from courier-imap. A really important feature for me from courier is that expunged messages end up in Trash where they are removed after 7 days. I can get an approximation of this feature as described in the documentation: namespace private { prefix = INBOX. seperator = . inbox = yes } namespace

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I''ve been unable to find anything definitive about what happens if I use RAID0 to join an SSD and HDD together with respect to performance (latency, throughput). The future is obvious (hot data tracking, using most appropriate device for the data, data migration). In my specific case I have a 250GB SSD and a 500GB HDD, and about 250GB of

Hi, If i set strict allocate = yes in samba to speed up the transfer of a mssql database dump, then btrfs does not compress the file. I have tried it also by just copying a small file in Windows to the samba share and the same. I have tried btrfs mount options autodefrag and then btrfs fi defrag -c and the file still does not get compressed. I have tried kernels 3.6.11, 3.8 and 3.10.7 on FC16

Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? I have the weirdest problem ever. I have three SIP endpoints. SNOM phones, if it matters. Their extensions are 200, 201 and 202. Apart from the username/password, the sip entries in sip.conf all have identical configuration. They're all NAT'ed behind the same IP. 200 and 202 registers

Hi guys! Thanks to your help I've my IceCast/LiveIce server up & running but I´m still having problems: My IceCast server is on the internal LAN and it's NAT'ed behind an OpenBSD IPF firewall. Here's my network setup: Internet ---- [OpenBSD firewall] ---- hub ---- [IceCast server] | | +

Hi! Sorry! I forget to mention that port 8001 it's open at the firewall too. So winamp should work fine with this configuration. Thanks anyway Kurt! Any other ideas? Marcelo Gulin <p>-----Original Message----- From: owner-icecast@xiph.org [mailto:owner-icecast@xiph.org] On Behalf Of Kurt J. Dreistadt Sent: Miércoles, 10 de Abril de 2002 06:43 a.m. To: icecast@xiph.org Subject: Re:

Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This

Dear list, I have a vector (array, table row, whatever is best) of frequency values for categories (or bins), and I need to find the median category. Trivial to do by hand, but I was wondering if there is a means to do it in R in an elegant way. The obvious medioan(vector) returns the median frequency for the binns, and that is not what I want. i.e,: freq cat1 1 cat2 10

On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > Just a guess (without knowing about your network), but are the two ends > points on public networks and visible to one another? If not the reinvite > may be passing an internal (nat'ed) address to the other and the connection > will fail...just a though t38modem -tt -o /var/log/t38modem.log --no-h323 -u 91 --sip-listen

All, If I have video phones behind an asterisk server (with 2 network cards) and all the phones have extensions. Internally everything works great. Now for people that want to call my video phones external to my office is there a way to do that? On the extenal persons phone enter an IP/EXTEN where IP is my server and not the phone? Can that work? Would I have to have PUBLIC IP address for every

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

I'm stumped. We have a CentOS 4.4 box with something like the following IPs: eth0: 64.61.61.100 eth0:0 64.61.61.112 eth1: 10.0.0.100 lo: 127.0.0.1 (of course) We want requests to 64.61.61.112:80 to be NAT'ed to 127.0.0.1:8080. This seems like it should be really simple -- maybe the following? -- I've tried variants combining this rule with a POSTROUTING rule, tried dnat,

So I have a freebsd machine setup, and it has 2 nick's in it. One is a REAL IP, and pulls in internet, the other is a a nat'ed nick, with samba running on it for the machines behind the firewall. Anyone know what to toss into my smb.conf to not let my outside nick xmit smbd/nmbd ? No one can connect via that interface, but people/viruses/whatever are still trying. Thanks! C.

Hi Everyone. I'm in a quandry & don't know which way to go. - Obviously I'm an Asterisk newbie although I've been watching this list for over 2 years now. I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running here at home. - It's on my home LAN - NAT'ed behind my LinkSys router. - On the same LAN I've got a Cisco 7940, 7960, and

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

In the setup I'm looking now there are IPv4-only and IPv6-only SIP-clients (or UA's) but the Proxies (or servers) are v4 and v6 dual-stacked (to make things a bit easier). I think there will be a lot of IPv6-only SIP-clients (e.g. mobile clients and all the existing IPv4-NAT'ed networks) and a lot of IPv4-only "old legacy" SIP-clients. We have to find ways to deal with such

On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: > > On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > > > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: > > >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > > >>> Just a guess (without knowing about your network), but are

Hi all, My Asterisk problem today involves getting a SIP client on a private net to register with a server somewhere else on the Internet. This worked for me about a year ago no problem, but now I see an error message on the remote server every time the client attempts to connect (the server is running Debian lenny with Asterisk 1:1.4.21.2~dfsg-3). Here's an example: [Nov 11

Hello! I'm still trying to get a working t.38 configuration w/ pjsip. I'm now able to send t.38 faxes to my own extension: hylafax -> t38modem -> extension -> extension -> t38modem -> hylafax. The fax is sent by t38modem. The receiving part of t38modem accepts the call, sends ReInvite for t.38 and things are working as expected. Now, let's do the nearly same

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=554 ------- Additional Comments From fhagur@gmail.com 2007-04-17 05:04 MET ------- I have been wondering about this bug and had similar problems myself here in my Debian system, linux-kernel 2.6.18 iptables 1.3.6. I too saw that some packets became transmitted illegally through the ppp0 interface, when they just shoudn't. What I