similar to: Good, affordable IAX hardphones?

I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable,

Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards

Hi, I'm interested in extending OpenSSH's PKCS#11 code to support ECDSA keys, but have so far been unable to find anyone who can sell me a smartcard that supports it. They certainly exist - AFAIK it's required by the US PIV standard, but obtaining cards that support it in single digit quantities seems all but impossible. Can anybody on this list help? I'd want 2-6 cards/tokens

Hi there, I know this question is kind of stupid. But, I don't know anywhere else to ask. I've received some answers when I asked about the need of having a zaptel interface to make the meetme application work, that said that it was better to have a real hardware then the zaptelrtc software modules. So, my question is, would any of the following cards work with the meetme

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

Hello All, I have been using the following phones with excellent success inside my lan: Cisco 7960 Polycom IP600 I have also been comfortable with the SPA-3000. I recently got a SPA-841 and the quality is aweful. Even in stock setup, it's like when I speak into the handset I sound like I am very far away. I want to get a hardphone for europe, but I need to be able to use a STUN server, and

I'm considering building a network gateway using a Via EpiaCN10000EG motherboard like seen here: http://www.logicsupply.com/product_info.php/cPath/79/products_id/444 I've heard that this is a pretty recent mobo and drivers for it are in Ubuntu but that a 2.6.9.x CentOS kernel might have troubles with it. Anyone have any opinions on this mobo? Thanks jed

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.

In trying to get my services up and running, I've encountered the usual spate of first-time issues. I was wondering if there was a good FAQ or Howto on troubleshooting NAT issues. The equipment that I'm using is typically a Sipura ATA or hardphone (SPA-942) sitting behind either a Linux box as firewall (using IPtables and Arno's firewall) or else a Cisco IOS device running 12.4T

Hi, I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client. Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I

Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the

Hello, I am working for a charity in the UK and I am projecting a new phone system. We would like to connect our two-wire telephones (40 or so) to an ADIT 600 channel bank, and connect that into an Asterisk box via the CMG card or T1 card. I have been in talks with Carrier Access about the purchase of a new channel bank and we tried to get a minor version of it first for testing with the

Hello There's a lot of information on VoIP at www.voip-info.org ... but there's also a lot of outdated information there as well :-/ Since SIP is a pain to use when NAT is involved, especially when both the Asterisk server and the remote phones are behind NAT... I'd like to try IAX to see how it works and if it solves the issue. I'd like to start with a softphone (Windows

For high-availability Asterisk when using PRI Lines, you might also want to check out our product (FSV-4PFS). It's available at www.failsafevoip.com Bill Also to add to the last post does this device have hardware echo cancelation? if it does it could be a great replacement, if not may not be what I'd really want to use. Thanks, tom I've been looking at the RedFone foneBRIDGE2

Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan

Hi list. I want to know if there is any console o secretarial hardphone that works with asterisks. I mean a phone in witch i can see the state of the extensions, the phone lineas, etc. Can hold o transfer easly a call, etc. Thanks Alvaro Parres

Hi, When a SIP hardphone is transfering a call while ringing (caller and callee don't speak to each other) using phone's Transfer key, it seems BLINDTRANSFER remains empty. Though I can see a 302 MOVED TEMPORARILY message coming in. Is there a work around or something obvious I'm missing (it's the first time I'm playing with Dialplan transfert features. context mylocal {

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

What hardphones support xml/html/xhmtl/microbrowser? I need an inexpensive SIP hardphone that can run simple applications (queue status, etc). The phones I know of: Aastra 480i, 9112i, 9133i (though limited by 3 LCD lines on the 91xx seems kind of silly) Cisco 79xx Mitel 5235 Polycom IP601 Any others? -Dan