similar to: WiFi phone radiation regulation?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin

Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details

Hi all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630

Anybody ever managed to implement a solution where one could forward a voicemail from one * server to another? Dominique

I've just seen this post: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is

Greetings, I purchased a WiSIP at the VON conference and am now trying to configure it to work with Asterisk. I have read all of the previous postings regarding the WiSIP and most of the information apparently does not apply to the version of firmware installed on my phone (version WF.00.0F). I cannot get the WiSIP to register with my Asterisk box. It leases an IP from my DHCP server, then

I recently got a Pulver Innovations WiSIP wireless SIP phone to determine if we want to use them in our organization. Since the WiSIP phone arrived last week, I have had nothing but headaches. I do think I now have the problem narrowed down. I have spent a bulk of my time trying to get the WiSIP to work with a couple of Linksys WAP11 Version 2.2. I have met with no success in getting the WiSIP

Hello - Here's my first impression review of the first SIP 802.11 phone. I got my hands on the "first" one sold, so that perhaps makes me the first person to have a real 802.11 SIP phone commercially in the US interworking with Asterisk. Whee! Can someone point me to other commercially shipping phones to prove me wrong?

Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with

Hello, I found serveral discussions about the Zyxel ePhone Prestige P2000W and the WiSip from Pulver Innovations on this mailings list but still have some questions: 1) are there other affordable wireless SIP Phones on the market? I haven't seen or found anything else till now ... 2) is p2000w and wisip the same hardware?? so could I use firmware from both companies regardless of what

Hi Folks, I have an IAX trunk link to a collegues house. I'm using AAH and he's got the latest CVS as of last Tuesday. Problem we're having is this; when I dial his extension 7201 (Pulver WiSIP phone) his * box sends me 1 ring and then Alison's busy message. If I call his 7202 extension (X-Ten Pro on a Win2K laptop) I get through but with only 1 way audio (me to him). Until

Folks, I'm looking for a SIP or IAX phone for field techs to take with them when out on service calls. The regular desktop phones are just way too big. Is there anything like the size of a full-sized cell phone? Or smaller, not I doubt that... Tim --

> "If" I understood your initial objective correctly (and I may not have), > the user's phones are negotiating the codec to be used for each rtp session. > > Asterisk parameters can be used to dictate rtp sessions between the sip > phone and asterisk, but that won't influence the next step in which the sip > phone negotiates a new rtp session directly with the

I'm fairly new to VoIP, and brand-new to Asterisk. I am wondering what are the best/least expensive phones or phone adapters to use with Asterisk, both wired and WiFi. I would appreciate any insight or recommendations you may have. Thanks, James arbaughj@myrealbox.com

Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer

I am having problems with * not hanging up an incoming PSTN line, if that line is not answered before the person calling in hangs up. The line hangs in various states, it has hung with a busy tone, with no tone at all. I am running *@home and have a digium 4port line card. This was configured by the genzaptel command I then added trunks for each line. I also have a Pulver WiSip phone which I

I'm not having any trouble with interrupts, but here's my /proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the SMP kernel (2.6.5-1.138). I don't think I need to worry about uhci_hcd, nothing is plugged into USB, but libata is the disk driver. How do I get libata and wctdm to use different interrupts? $ cat /proc/interrupts CPU0 CPU1 0:

Looking for WLAN - WiSIP - WiFi phone recommendations and experiences. What works, what doesn't. Thanks. Jim James H. Thompson jht@lava.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040716/2e0d45a5/attachment.htm

A while back, I asked about using Asterisk in a medical environment where the task is to write a program that connects to a phone and sends a message like: Hello Mrs. Jones. How are you doing today? Press 1 if you're OK. Press 2 if you need help. Or start talking, and your message will be passed to a person. After connecting and sending the sound file, the program would

_Description_ We are looking for an expert LAMP (Linux, Apache, MySQL, Perl, and PHP) developer with some Asterisk experience who is based in Western or Eastern Europe or Asia. We can work with an individual or an organization. You must be fluent in English. We need you to help expand development of Signate's core software products. As part of the Signate development team, you will