similar to: Yuxin hardphones feedback

Good Evening from Toronto. I have just installed Asterisk@HOME 2.6 and I am having difficulty with getting the YUXIN phones that I am using to work with Asterisk@HOME. I have flashed the units to 1.50 from the YUXIN Home Page. Does anyone have any history with these phones and Asterisk? Does anyone have the proper settings to get them to work with Asterisk? Am I using the proper firmware

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

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:

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

Hello Since SIP/RTP is a pain to use with road warriors who need to connect from any location over the Internet, I'd like to get them some IAX phones instead. For those of you using this protocol instead of SIP, what would you recommend as IAX hardphones and Windows (and ideally Mac) softphones? Thank you.

hi all anyone that've heard of any working IAX hardphones yet? roy

Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next

I'm looking to purchase and deploy a bunch of hardphones for agent use. The phones will have to register with Asterisk and/or SER, depending on where the phones go. They need only one line, G729 codec, and no super fancy features. Preferrably something that is easy to provision. I would think the BudgeTone would be good, but then I've read so many people complaining about them, and some

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

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

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

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 SIP is such a pain to use when NAT is involved that I'm willing to buy an IAX hardphone for someone who works remotely over the Net and needs to get calls from our Asterisk server, itself behind a NAT. Are there good, affordable IAX phones you would recommend? Thank you.

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 {

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

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

Hi, I have used the Budgetone 102's extensively on Asterisk and found then quite reliable as long as you update the firmware. The GXP2000 is quite a mess at the moment as the current firmware does not support 3 quarters of the advertised functions and codec support is extremely limited. I have tested the unreleased latest firmware update for the GXP2000 and it's an incredible difference

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

Like most others on this list I also have some really annoying echo whenever a call goes out to the PSTN from a SIP phone... SNOM/Budgettone -> Asterisk -> X100P -> PSTN I have tried every echo canceler in the makefile and turned on and off aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and I can get it reduced to only a few seconds on the intro of the call and

I have been testing * with some Cisco 7912G's, in hope to trash our Nortel system. One feature our Nortel system has that I will need to fiqure out on the * system is paging. Is it possible to page a group of phones (all phones) with announcements? We are a k-12 school and we use our current phone system to make announcements on the phones monitor speaker. Any direction I can be pointed in