search for: 20clients

Displaying 5 results from an estimated 5 matches for "20clients".

2017 Sep 07
1
Firewalls and ports and protocols
...lusterFSonCentOS we only need port TCP open between 2 GlusterFS servers: Ports TCP:24007-24008 are required for communication between GlusterFS nodes and each brick requires another TCP port starting at 24009. According to https://gluster.readthedocs.io/en/latest/Administrator%20Guide/Setting%20Up%20Clients/ we also need to open UPD: Ensure that TCP and UDP ports 24007 and 24008 are open on all Gluster servers. Apart from these ports, you need to open one port for each brick starting from port 49152 (instead of 24009 onwards as with previous releases). The brick ports assignment scheme is now complian...
2018 Jul 18
2
GlusterFS
Hi! For the files that are in the share Regards; On 18-07-2018 18:09, Micha Ballmann wrote: > You mean your Samba shares? For fure. > > Regards > > Am 18. Juli 2018 22:54:28 MESZ schrieb Carlos via samba > <samba at lists.samba.org>: > > Hi! > > I would like to know if I can use GlusterFS to replicate the files in my > samba domain member?
2004 Sep 28
3
CODECs and sip.conf and voice quality
Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the
2004 Dec 16
3
Get asterisk out of the RTP stream?
Here is the setup: Phone A (in NYC) on own bandwidth. Phone B (in LA) on own bandwidth. Asterisk box in Houston,TX on own bandwidth. Both phones contact asterisk to register. Not much bandwidth used for this as it is a few packets every hour or so. Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk calls phone B. Both phones are connected and both people are talking.
2004 May 25
2
sip phone problem
Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem