similar to: Lower cost router suitable for VOIP ?

Hello all, this is not really an * question but it is somehow related, i am trying to develop a working proposal for cheap and quick telephony services using Voip running over *. By running a wireless network (over 802.11 a/b/g devices), i plan to be able to reach customers directly with eithe table top or handheld 802.11 sip enabled phones. But the disadvantage is that how do i power each radio

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

Hi, is there some easy way how to shut down vmware ESXi host? We have same apc smart 1500 and I passed usb port into one virtual linux, so it can monitor ups state via NUT. So how can i make esx server avare of low battery? Maybe using vsphere-cli utils? With regards Libor -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hello everyone.... Searching the archives and google always comes up with entries regarding the "dyn" dns option in the 7960, but I can't find answers to my specific question.... My 7960 is connected via cable modem and is NAT'ed (everything is working fine). On the 7960 under SIP configuration\NAT Address I have the public IP of my cable connection. Comcast gives me a

Hi, We're installing an * box next week (pbxtra from fonality) and I'm trying to come up with a solution for remote users that want a phone in their home. I need VPN and QoS capability, wireless support would be a nice to have. Ethernet handoff is fine, i don't need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces

As seen on my post at: http://www.sveasoft.com/modules/phpBB2/viewtopic.php?p=28112#28112 This works very well... It does NOT work with stable 4.0! sveasoft will be issuing a bug fix for this (4.1) in the near future. Final Rev of working script w/ asterisk support I'm not going to run alchemy on production machines until it is stablish. Remember to set your uplink properly and to set

current setup SIP phone 192.168.1.30 --> linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)--> (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address.... or #2 asterisk trying to get back to me as 192.168 on public internet.. got

Hi all I have succesfully set up PPTP VPN access to my company using PoPToP/pppd (thank you Tom for excellent documentation), everything works fine so far. Problem is security. As I understand it, since PPTP ports trough tunnel are opened to whole Internet (have to be), access verification is now based only on username/password combination. Can level of security be raised with MAC verification?

Hi, I'm trying to implement dynamic routing of incoming calls to local extension if previous outgoing call was unanswered. But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to 's-NOANSWER'. I guess this is normal, but I don't understand why ? How to workaround on this one ? Thanks in advance, regards, Rob. [outbound-capi-ISDN] exten => _0.,1,NoOp(Calling ISDN

> Chris, > > I believe it would be nice to send the info also to the list. So > others would be able to benefit as well. You've got at least 2 people > interested :) > > Marconi. Ok, I just wasn't sure because it's more of an 802.3Q/Routing issue rather than an * issue, but if everyone's cool with it I sure will... I'm not the only one who knows this

I've looked around briefly for what options are available for encrypting the media stream using asterisk. I did not see any SRTP support, and it looks like there is some initial work on iax2 encryption, but whether it works is still open for question I guess. I'm also curious of other solutions that could be bolted onto the front end of asterisk to provide encryption, and are there

I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from

Running * with default config files except for sip.conf. Any call made is dropped 5 seconds after connection, with the following messages: Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar

Hi All, I'm looking for a good voip hardphone that has a decent set of the "regular" features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel

Sirs, I have recently been interested in your Maxstat. I have computed with my own programme the ranks (by using the Kaplan-Meier method and the log-rank test) with the formula (Observed-Expected)/(SQR Var). The results are similar but not exact to the M value obtained with the Maxstat. I would like to know whether you are using some correction or adjustment in computing the different ranks. Thank

Did anyone see the article? It''s the first time I really noticed that these little Linksys routers are such a fully fledged linux machine with a decent processor and a replacable firmware. I am now itching to get one to replace the multipurpose firewall desktop machine. Has anyone experimented with the current state of the firmware and how advanced you can get with tc rules? For

Hi, we want to upgrade our infrastructure and are considering renting two different types of server by a provider: - A: one is with 80 GB SSD (and 12 GB memory) http://www.ovh.co.uk/products/eg_ssd.xml - B: the other with 750 GB SATA2 (and 8 GB memory). http://www.ovh.co.uk/products/eg_best_of.xml The target infrastructure will host: - a few public web sites (PHP / MySQL) - a development

Hello! Over the last couple of days I?ve been trouble shooting a strange problem with Asterisk. First of all, I should say that I?ve been running Asterisk on a Fedora Core 3 box since last May, but decided to do a reinstallation of everything due to some problems we?ve had with echos during conversations (100% SIP based, so no ZAP echos). We are talking about a low-volume installation

Hi, not sure if it's possible to do something like this. My main * box here at home is sat on the local lan, 192.168.1.0 , also on this lan I have another * box which is running as part of a wireless Meshbox (http://www.locustworld.com) . Now then, this meshbox has a wireless connection to another meshbox at a friends house. These meshboxes use weird IP addresses in the 1.x.y.z range

Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090817/cd8c6546/attachment.htm