similar to: Uniden uip200

I just got my wife this phone, and when calls come in on the queue it will not ring, if I dial the ext it rings, here is the config from the tftp server: Thanks unidencom.txt # UIP200 Mass Configuration System Generic File # Notes: # 1. Lines start with '#' are comments # 2. To leave a field value unchanged (as saved on local phone), leave value to blank. # 3. To set a field's

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

Hello We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since. We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: It does show up in asterisk a few seconds after the UIP200 reboot: -- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200 but

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

Hi, I've read through the archives, and wanted to get an updated opinion on the Uniden UIP200 phone. Seems like there were a lot of opinions that it was a good phone, but there were a few items that people were waiting for firmware updates for, but that was in 2004. I'm going to be using them in an office, 12 phones, on a LAN connected to an asterisk box. Thanks for any advice or

I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in

Hi, all. Got my Uniden UIP200 today (ordered from thetwistergroup.com), and was very excited to set it up... until I came to the realization that there were no docs with it whatsoever. There was, however, a sheet of paper with the stock warnings (don't use the phone in the tub, etc.), AND a URL -- for documentation. Score! Well, no. The URL it gave me, bcs.uniden.com, does indeed have

I guess this is an FYI Running NMAP (protocol detection, common ports) against my UIP200's causes them to reboot. firmware BS4.56. Be careful where you point that thing. -Nate

Anyone using the UIP200 with *? I am having difficulty getting the phone to register and *stay* registered for more than 4 seconds. The * console shows the UIP200 registering and records the user agent. The UIP200 displays station name and time on its LCD, then 4 seconds pass and it shows #1 DISCONNECTED. The * console reports no problem and believes the device to still be registered. A call

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

I am trying to get a new Asterisk installation running using a Zultys ZIP2 phone and a Uniden 200 phone. I have the system working reasonably well (although probably not optimal) except for a couple of items. First, I can't get the voice mail message light to work on the Zultys phone but it works just fine on the Uniden phone. Second, the time presented on the Uniden UIP200 phone is 1 hour

My users are reporting that some voicemail messages are being cut off in the middle of being played back. The recordings are OK (they play fine when forwarded to e-mail, and they can often be accessed OK during a later call to voicemail). I found nothing in the archives on this -- ideas anyone? RH9, P4, CVS-HEAD-09/02/04-08:44:34, aggressive echo suppression turned on. Jim Shilliday IT

Anyone used any of the Uniden VoIP gear, any comments one how easy it was to integrate with asterisk? Also quality of the gear in general? This unit _looks_ pretty nice on paper http://www.uniden.com/productpop/00_productpop.cfm?prd_code=UIP1868 -- Harondel J. Sibble Sibble Computer Consulting Creating solutions for the small business and home computer user. help@pdscc.com (use pgp keyid

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

I have a UIP200 on the way for eval. Does anyone have tips or tricks to get it working right away with * ? I hate having to go through the pain someone else braver than I went through already. :) Tim --

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

Phone won't register on LAN port registers but doesn't work on PC port. SIP to SIP works. Anyone have a Configuration that works out there? Phone has 4.63 Firmware Asterisk@Home 2.0 Beta -------------------------------------- Jeff Herring / jeffh@sldsi.com Seacoast Laboratory Data Systems, Inc. Voice: 603 431 4114 x14 FAX: 603 431 2112 --------------------------------------