similar to: Uniden UIP-200 Multiple line appearances

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect'

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

I just purchased 30 of these after testing one for a few months and would like to quickly purchase another 40. We really like these phones: good sound quality, good echo control (no echo in speaker phone), power over ethernet support, 10/100 switch, 8 programmable keys. Unfortunately we missed on big problem with Call waiting in our testing. When using asterisk 0.9.1 or rc2 the phone will reboot

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

These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

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 --

No kidding, every time. I know I have the config via tftp working. Funny story - I was getting nowhere with it and then decided to tcpdump on the tftpd box, and wow! The UIP-200 tftp client was looking for the uniden<mac>.txt in lower-case! Hah! That was easy to fix. Now the config is transferred to the UIP-200 at startup. It registers to the * server. The phone displays time and

I am looking for a large number (probably about 100 or so) low-cost phones that I can hang on the wall. I need the phones to use PoE. Do the Uniden phones support wall-mounting? These phones are not going to be high-usage; they simply need to be there in case of an emergency. Another question, along the same kind of lines, has anyone figured out how to keep the SoundPoint IP 600 receiver

We just did an upgrade on our Asterisk to the CVS version and our Mediatrix 1102s stopped working correctly. Our Asterisk is connected to the PSTN with a PRI. Calls from the PSTN to the Mediatrix 1102 work fine. The issue is calling out to the PSTN from the 1102. Asterisk looks like it process the call just fine except there is no talk path. Get this, though: If you flash hook and then

I have several too and I also see this problem on occasion. Like you say, it is fairly rare and I can't pinpoint a cause or tell if it is a symptom of something else. I think I wrote to tech support about it but never heard anything. I'm wondering how long they will continue to support the phone. -Nate > -----Original Message----- > From: Jerry Geis

Anyone had experience with this phone? The interesting features that caught my eye were the RTP encryption, speaker "pager", and built in 4-port ethernet switch. Of course, RTP encryption doesn't exist in * yet, but it might be interesting (I've had people ask about it, but I don't know how serious they are about needing it.) http://www.zip4x4.com/summary_ZIP4x4.htm

Hi all, Does anyone know of a method of sending a raw G711 stream to an address in Asterisk. For example, an application that takes a argument of a phone and a port. The reason? I have found a method to paging on Zultys ZIP2 and ZIP4x4 handsets. Basically it involves sending a stream of RTP data to port 3771 to multicast address 224.0.0.1. Would it need to involve me writing my

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

Just wondering.. Any experience with the UIP1868 ? I assume that it can handle a single SIP line (can't seem to find the manual at their site :-).. They mention also T38 in their webpage.

Does anyone know how (if possible) to do three way calling on the UIP-200? There doesn't seem to be much info about this phone, but all the feature lists I've read says it can do conference calls. I can't seem to do it, though. Any help would be appreciated. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us

Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214

The UIP(update in progress) is set when RTC is in updating. And the update cycle begins 244us later after UIP is set. Signed-off-by: Yang Zhang <yang.z.zhang@Intel.com> diff -r 47cb862a07c2 -r edc35b026509 xen/arch/x86/hvm/rtc.c --- a/xen/arch/x86/hvm/rtc.c Mon Mar 05 14:39:07 2012 +0800 +++ b/xen/arch/x86/hvm/rtc.c Mon Mar 05 14:39:41 2012 +0800 @@ -28,6 +28,8 @@ #include

And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw

The piece of plastic is built in to the IP600. Use a screwdriver or something similar and push the little piece of plastic out from the top inside the cradle. It will pop out, and you can turn it over and reinsert it upside down to hold the receiver in place. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

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