similar to: Multiple locations, 2 servers - planning questions...

>It will be about 100 phones at about 20 locations all within >about 4 miles of each other. Perhaps a more pressing question might be how you are going to backhaul Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100 metres reliably, and using Ethernet repeaters every hundred metres or so isn't practical. You will need a fiber backbone or something like that. What

Seeing as IPV4 is near it's end of life (http://www.internetnews.com/infra/article.php/3915471/IPv4+Nearing+Final+Days.htm), I'm curios as who know whether everyone is ready for the changeover to IPV6? Is anyone using it in production already, and what are your experiences with it? -- Kind Regards Rudi Ahlers SoftDux Website: http://www.SoftDux.com Technical Blog:

I am looking into supporting around 20 SIP clients (ATAs, IP softphones, etc) distributed in around 10 different end points (in South America). For the most part they all have narrow band connections 64kpbs, 128 at most and I?d like to use g729 all around (don?t have too many alternatives) To start with, I will have one * with no gateway to the PSTN and eventually a few * boxes with termination

Anyone out there using small-midsized (2-4 TB) SAN solution among multiple Asterisk systems? I don't have the budget for an EMC-caliber solution, and can't seem to find much else out there. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent

# === The post relates to Dovecot v2.2 running on AWS Linux v1 === # Hello all? The Dovecot sockets on this server are installed in two different locations one of which is different from what I have on my other Dovecot installations which run on Centos-7. Here are the sockets paths: ? all sockets EXCEPT "auth-master? ?> "/usr/local/var/run/dovecot? <<<<< this

I have several branch offices all running Asterisk PBX's that register to each other via SIP so that calls can be transferred from office to office. Everything is working great on the office to office transfers, but I'd like to somehow make the CallerID more useful. Currently if an extension at Office1 dials an extension at Office2 the CID on the phone at Office2 says

I'm looking at setting up Asterisk for a completely IP environment. All intercompany calls. I work for a ski area. I currently use a 3Com Superstack for in our office. And an old small town phone system for up at the mountain. The phone system is dying and I'm hoping to bring IP to replace the old phones. It will be about 100 phones at about 20 locations all within about 4 miles of each

OK, I've been searching the archives and google with little luck... I have a directory which needs to be shared with a number of UNIX clients via NFS and with Windows clients via Samba. What is the best way to do this? Should I configure Samba to share the actual NFS mount? Or should I configure Samba to share the actual filesystem directory? Reason I ask, we were using the former method

Hi, I need to connect 2 asterisks in 2 different countries (A and B) for one company so it's possible to make connections between the 2 offices. For connectivity reasons (NAT traversal) i want to connect the 2 asterisk with IAX so that when a user on office A connects via SIP to user on office B the call is going trought IAX channel. Can anyone give me an ideia how to accomplish

Hi, I need a feature from asterisk with dahdi channels, if there is an incoming call, it should ring on several dahdi channels. My channels look like: OFFICE1=DAHDI/13,,rtT OFFICE2=DAHDI/14,,rtT If I add this line: exten => 12345678,1,Dial(${OFFICE1}&{OFFICE2}) only OFFICE1 rings. If I change it to exten => 12345678,1,Dial(DAHDI/13&DAHDI/14) DAHDI/13 and 14 rings together,

Hello, I have 2 offices connected through VPN (all ipv4 and ipv6 traffic allowed), every office with it's own subnet. I built a DC in office1 for mydomain.local, built a second one in same office and joined mydomain.local with no problem. Then i built a DC in office2, but when i try it to join mydomain.local, the join process blocks at "Setting account password for OFFICE2-DC$" and

Hi Folks, First of all, I've been reading a lot and a lot about asterisk and ISDN-BRI, but is really a complicated world, not like PRI or Analog. I have 2 offices with 2 NT boxes and I want to connect them with Asterisk to route outgoing from one office two the other. At the end I want them to look like this Office2 will have only VoIP phones (not ISDN phones connected to a card in asterisk)

Hi, I was thinking about a common (I hope!) use case of Glusterfs geo-replication. Imagine 3 different facility having their own glusterfs deployment: * central-office * remote-office1 * remote-office2 Every client mount their local glusterfs deployment and write files (i.e.: user A deposit a PDF document on remote-office2), and it get replicated to the central-office glusterfs volume as soon

I think I have a working dovecot installation, and the time has come to migrate my users. There are only 4 of us, so it can be done manually per user. I looked at this page in the wiki http://wiki2.dovecot.org/Migration/Dsync but I am totally confused. 1st off, it uses doveadm, but with commands parameters that dont match my doveadm's man page. In fact they look more like dsync

I need to setup incoming (over an 800 number and some local DID's) and outgoing phone calls (all over the country) with an Asterisk server. This asterisk server has 20 Polycom 430 phones connecting to it. I need the best possible SIP provider out there. I have tried http://www.nufone.net and http://www.broadvoice.com and they do not even come close to the expected quality. Does AT&T

I am trying to trouble shoot one of my ISP's network and compare to my other ISPs offering. Although network 1 is reasonably fast and has low enough latency, voice quality is not good and the reason for this is not readily apparent using standard network tools. What tools can be used to assess the quality of the network in terms of it's suitability for voice? I am using ping, mtr,

Dear list, I?m trying to setup dovecot/postfix/getmail for a single user application: I have 3 email accounts and 2 main computers (home and work). I want to have an imap server on each machine with sync/replication so that read/reply/delete status and sorting into folders are identical. This part is achieved via dsync and seems to work. Messages are fetched from my ISPs with getmail on both

I have several branch offices, each with their own Asterisk server (version 1.4.22.1) handling their PBX functions. All of these offices need to talk to each other. In sip.conf I created a peer entry for each office with a username of branch-user and a friend entry for every branch-user with the username being just the branch, for example: [Office2] username=Office1-user host=10.10.80.253

in a recent post, sven made a clear/cogent arg for sdbox vs maildir > While Maildir seems to be an obvious choice, it has some hidden > performance impacts, namely it relies heavily on information encoded in > the filename, which results in a big overhead in meta-operations, > because Dovecot needs to stat() every file in a directory to gather > those information. i've

How ISPs May Quietly Kill VoIP Communications Posted by Zonk on Friday March 18, @10:59PM from the poor-kids-never-had-a-chance dept. ravenII writes "PBS's i'Cringley's informative piece gives an eye-opening look at the anticompetitive behavior of some ISPs who are showing up late to the VoIP game. This is not something that could be easily mandated, and the beauty of this