similar to: UIP-200 conference call

I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO

I've come across an odd dialing problem with my * setup. After * has been running for a while, if I try to dial out on any of my zap channels, (both are X100P cards) it picks up the line but never sends the DTMF. Has anyone heard of or seen this problem before? Right now I'm looking at 19 hours of uptime on * itself. If I restart it, everything works fine for a little while, then the

Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the

Because I'm tired of seeing Debian users look like idiots for using such horribly out of date versions just because they run "stable", I've set up a page with .deb packages of my 1.1.x rebuilds for use under lenny. http://www.rollernet.us/opensource/ Standard disclaimer applies: use at your own risk, no warranty, no support, if it eats your first born or erases your computer

I can't seem to make the "Hold" button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press "Hold" button 3) Hang up phone What I expect is for the call to go on hold until I pick up the receiver again. (Like my SPA-2000's, except it's a flash, but I can hang up the phone and the call waits there for me to pick it up

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

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."

On Jun 20, 2004, at 9:16 PM, asterisk-users-request@lists.digium.com wrote: > I am new to Asterisk, but though that I would need calls being > answered in > different contexts. > > How can I direct one line to a given context and the other one to > another, > or is there a better way??? > I use this at the end of my zapata.conf file to do exactly what you're

I'm having an echo related problem with Zap channels that are answered before a dial takes place, such as for IVR menus or fax detection. Basically, it sounds like the volume gets turned up to maximum while * is ringing the internal extensions waiting for someone to pick up, so when you pick up the phone to talk, you get an over-amplified echo. The sequence is like this: exten =>

Hi all I am in the process of migrating Courier to Dovecot, am using courier-dovecot-migrate.pl<http://www.dovecot.org/tools/courier-dovecot-migrate.pl>it works fine. I want to migrate each user as they login using post-login script (As i have huge no of users). We are using NFS storage, users are authenticated using mysql, User home directories as passed using variable mail ( usign mysql

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

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

If you want to run Dovecot on Debian Lenny for a Production System (with Fast Upgrade System for Security Patches), What would you recommend? Stick to the current port with Dovecot Version 1.0.15 (1:1.0.15-2.3) Use Backports Version 1.1.13 (1:1.1.13-2~bpo50+1) or Use any of the packages from:

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 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 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

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 --

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

I just installed CentOS 5 + Postfix + Dovecot. Now When I send mail via telnet from one user to another via Postfix (MTA), I have no problem. The recipient gets the message perfectly. Now this tells me that Postfix is working fine (I assume) but then I configured Dovecot to use IMAP4 and then configured my Thunderbird client. Now all the messages I previously sent using Telnet showed up in

Dear WizaRds, I would like to fit a curve to ten points with nls() for one unknown parameter gamma in the Kahnemann/ Tversky function, but somehow it won't work and I am unable to locate my mistake. p.kum <- seq(0.1,1, by=0.1) felt.prob.kum <- c(0.16, 0.23, 0.36, 0.49, 0.61, 0.71, 0.85, 0.89, 0.95, 1) ## how to find a function that fits these points nicely? plot(p.kum,