similar to: Hardware PSTN Gateways?

I have a couple of Xen 3.1 servers running on Ubuntu 7.10 AMD64 Server machines. I decided to to upgrade these machines up to the latest release of Ubuntu (8.04 LTS) which upgrades Xen to version 3.2. I upgraded the first machine, installed the xen component. these machines all have two network interfaces one connected to the WAN and the other to the local LAN network. In a tutorial i read a

I have a system with a large LVM VG partition. I was wondering if there is a way i could share the partition using nbd and have the nbd-client have access the LVM as if it was local. SYSTEM A: /dev/sda3 is a LVM partition and is assigned to VG volgroup1. I want to share /dev/sda3 via nbd-server SYSTEM B: receives A''s /dev/sda3 as /dev/nbd0. I want to access it as VG volgroup1. I am

when performing a live migration i know the contents of the ram is copied over two the new xen server, but how about the contents of the swap partition. currently I have the swap partition in a seperate loopback file from the root partition. if i want to do a live migration, do i have to give the new server access to the swap partition file along with the root partition file? of can i just

>> I thought the W clients (both W7 and W10) reverse dns PTR records >> where added by the ADDC automatically. >> Yes, correct if your setup is correct then yes this works. With a default windows install and with a few GPO changes. > Early samba4 versions does not support (or have a buggy support) for auto->register PTR recors, as far i remember. I use windows to let

Hello, I hpe you can help. I have created a botable thumbdrive that allows me to update different bios's using the memdisk module. I have no problems loading the img or ima files on any equiptment other than IBM. I have tried on T43, T60, and T61, no luck. works fine on hp dell and all others tried. Every time after it loads the image it hangs after the message "Loading boot sector...

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt

On 2016-11-07 03:06, Rowland Penny via samba wrote: > On Mon, 7 Nov 2016 09:51:50 +0100 > "L.P.H. van Belle via samba" <samba at lists.samba.org> wrote: > > I thought the W clients (both W7 and W10) reverse dns PTR records > where added by the ADDC automatically. Yes, correct if your setup is correct then yes this works. With a default windows install and with a

OK, i am hoping that someone will be able to help me out. I am using FreePBX 2.8.1.4 I have two asterisk servers connected with a iax trunk. The trunk is working fine when used via the outbound route setting. meaning an extension on one server can call a specific extension on the other server. now what i want to do is set it up so that an incoming call (from a third server) is redirected to the

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

Hi, My setup: RedHat 7.3 PDC server with samba 2.2.7 rpm rebuilt with max connections patch and ldapsam and a few other minor changes, openldap 2.0.23-4. Windows 2000 Terminal Server with SP3, with various pre SP4 updates too. Various Win2K Pro, Win95 OSR2 clients. [homes] comment = Home writable = yes valid users = %S nt acl support = no oplocks = no path = /md3/profiles/%U

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

Let''s say I''m building a "to-do" list application using Ruby on Rails. I decide to sell it to a customer, and distribute the source code files to him. Can I then restrict the user from distrubting the files by creating a license that prohibits distribution? That is, is it possible to create a "single-user" license? Or, am I limited to distributing the

Hello, are the gsm codecs in Cisco as53xx or 36xx (gsm, gsmfr) copatible with Asterisk's gsm codec? Thanks in advance, Thomas

Apologies if this is an FAQ, I wasn't able to find an answer googling: Will any of the public PSTN/VoIP gateway services (Vonage, Packet8 etc) interoperate with * ? I'd like to deploy a box which provides PBX service for analog handsets, and handles inbound/outbound calls via both analog PSTN lines, and, say Packet8 VoIP service. I understand that I can do this by connecting the analog

So what are people using these days for SIP or IAX to PSTN gateways. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? 2. What about latency and reliability? 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if

OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a "w" in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w5555555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping

Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%

Le mar. 20 ao?t 2019 ? 14:30, L.P.H. van Belle via samba <samba at lists.samba.org> a ?crit : > > Hai, > > > > > In short. My network design previously work with Debian Stretch > > Servers and clients and some Windows clients (not many). > > > > Debian Stretch use Samba 4.5.16 so there is no unix_primary_group > > option for the clients. So I have

Hai, If the primary domain is set in windows, which is after domain join, it used that. Ipconfig /all and see primary DNS suffix. The dns suffix and first dns search list should be the same. Yes, other settings are possible, but stick to this for now. The Primay DNS suffix is used for the register of the IP in the DNS. The DHCP Service User MUST be a member of the DNSAdmins. The DHCP

Hi, I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI. Everything works fine with one exception. I seem to be unable to figure out why I cannot hear PSTN intercepted announcement ("number is not in service" etc.) when I'm calling a disconnected number through