similar to: Server load capabilities

We are using rsync to push DNS records from one machine (running rsync 2.5.6 as a client process on Debian woody) to a collocation server ( running rsync 2.5.6 as a daemon on RedHat 7.3). To test this process we are using a script that pushes one file from the Debian box to the collocation server. It appears that a file transfers, but upon checking there is no file transfer and an error in the

I'm proposing to install an Asterisk PBX at a collocation facility for a remote customer. Each of the customer locations will have an SPA-3000 with the FXO port connecting a POTS circuit and the FXS port connecting a fax machine or red phone. In addition to voice traffic, the customer has a high volume of incoming and outgoing faxes. Would it be possible, using g711 between the SPA-3000

i'm trying to set up outboud caller ID on 3 polycom ip 500 phones. here is how i have them set up. each phone is using 2 lines, all registered to * phone 1 line 1 is registered using the DID, and the next 5 lines reg to 03152 03153 03154 03155 03156 from my understanding, the way the polycom configs are set, i have to put the 03152 in the reg.x.address field. and it seems that is where

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike

Please add to mailing list. Thanks, LAUREN CANHA Staffing Coordinator IronPort Systems, Inc. 950 Elm Ave. San Bruno, CA 94066 Office: 650.989.3179 www.ironport.com <outbind://221-00000000FB683FD50A72FB408BA2313DEC2CE3190700406842A96543C 843ACB6B4ECDF016270000000BC63670000406842A96543C843ACB6B4ECDF01627000000 0C880160000/www.ironport.com> - The Leader in Email Security

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

Hello All, I?m on the technology committee for a fraternity at the University of Illinois. We?re looking into moving from our current ?party line? (one line shared between every two rooms) system to a PBX with voicemail in an effort to lower our monthly phone bill and provide better communication services. We?ve pretty much settled on Asterisk as we do not wish to rewire all of our pots

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown

Greetings all. I''m using the Microsoft "NorthWind Traders" database "orders" table as test data for the Rico LiveGrid JavaScript. The problem I''m noticing is that some of the records in the "ShipName" field contain ampersands -- which for some reason are not being escaped by Rails. Rails does in fact automatically escape all of the other

Dear all, I need to test for the existence of an index on a table. This cannot be done with sqlPrimaryKeys as it is not a primary key. Therefore I select directly from the systemtable of SQL-Server 2000 named sysindexes. This works well with RODBC Version 1.2-5 but not with version 1.3-0. Here is the code of the test example: sink(file = "proto.txt", append = FALSE, type =

I'm having a heck of a time getting vsftpd to work properly. When Iptables are OFF, it works fine, and when iptables is on, it dies. When I try ftp from a command line, here's what the session looks looks like: [root at mylaptop ~]# ftp ftp.server.com Connected to ftp.server.com. 220 Welcome to My Company FTP 530 Please login with USER and PASS. 530 Please login with USER and PASS.

Hello, I''m trying to configure several wireless routers with QoS support. The idea is to implement differentiated services so that VoIP traffic gets the maximal priority, then video, control traffic, interactive data traffic and best-effort traffic. I have seen that CBQ used to be chosen as qdisc for implementing bandwith share in DiffServ but now HTB is preferred because it is

Greetings, This is probably some configuration issue, but for some reason my system has stopped playing a ringing sound when an extension is dialed. The phone rings but there is no ring sound in the ear piece. Gene Kochanowsky

David, I guess I'm a little confused here. Are you asking me to provide a "pencil copy" of an LOA for your review? I don't understand why you need an LOA from us. We need an LOA from you to order circuits that will be billed to us that will be attached to your CFA. It was also my understanding that you had an LOA ready to be given to us, which had already been reviewed by

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

Hi! I am learning tcng without having experiance of tc and I am trying to build something that shall schedule traffic dependent on the value in the IPv4 packets ip_ttl field. I have read the tcng reference manual and cannot find information about forwarding. Is it possible to farward packets from ingress to egress without sending them upwards in layers?

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

Hi all, I''ve got XCP 1.0 up and running nicely and would like to use it in production. However I''m struggling with the concept of bandwidth management. It seems like such a common problem that everyone must have, but I can''t find any clear direction in which to go. The dedicated host I am using (Hetzner) gives me a 5TB monthly bandwidth quota which needs to be shared