similar to: Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore)

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

I have an Asterisk 0.4.0 install working with two grandstream budgetone 100 phones, gnophone, and kphone. This is a private network segment (172.17.x.x), with the PBX configured on my outbound firewall which has a public address (66.x.x.x). - I can make calls between phones - all extensions are working. - I can make IAX calls to IAXTEL. No problems (apparently gsm only) - I can call SIP phone

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

Hi Folks, Since there isn't a grandstream forum AFAIK I guess someone here may be able to shed some light on this. Apologies if this is viewed as offtopic.. I think I may have killed the firmware on my Grandstream Budgetone 101. I found a source for the 1.0.5.30 firmware and made the files available over tftp. The phone downloaded the files but now doesn't boot and hangs with a blank

I just got a Budgetone 101 and I have it hooked to my * box. I thought I'd read somewhere that we can program the buttons on these phones to send DTMF tones, thereby effectively programming them. However, according to the user's manual, they have predefined SIP functionality. My dialplan implements the festures I want (transfer, message, stuff like that), so for uniformity, I'd just

Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:daniel.andre@iris-tech.fr) IRIS

I know there are several folks hanging around the list that resell VoIP products, I'd be interested in getting your info off-list as I'm trying to persuade someone to get me a budgetone 101 for Christmas ;) I've checked the yahoo site, and they seem to only carry it in white.. Ideally, I'd like blue, so if you have it, please let me know. Also, I've been watching the list for

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

Hi, not sure if it is against the rules to sell second hand equipment in here but haven't seen anything against it so here it is. I'm upgrading to 2 lines so I have some spare equipment for sale here. This is an ideal starter pack and will get you going with 1 line and 1 extension. 1 x http://www.digium.com/downloads/product_sheets/X100P.pdf 1 x

Maybe I'm loosing my mind but I've just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. I'm using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone I'm missing? Cheers, Dean -------------- next part -------------- An

Hi there, I'm having problems with the Grandstream Budgetone 101 on hangup - "show channels"/"show channels concise" output is still showing the call's channels as active. The problem does not exist when I use SJPhone, so I'm assuming it isn't an Asterisk configuration issue. Has anyone seen this, or better, does anyone have a fix? :) Thanks, David. --

Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used

Hi all, I'm having a really frustrating time with a bunch of BT-101 phones. They've been trouble-free and working very well for the past several months. A couple of days ago, some of the phones (but not all of them, yet) have started acting very strangely. All phones are running firmware 1.0.6.7, and are identically configured (except for the user/authenticate/password things) on both the

I have a Shipment class that is used to ship packages between Branches. However, when I try to have two belongs_to in Shipment called shipped_to and shipped_from I can''t seem to make it work. I can get it to display the form properly but when I select the branches and click save I get the following error... "Branch expected, got String" Any ideas? I have added some code

Hi, Just wondering what people use for defect tracking generally in the RoR community? Is there an open-source/free online defect tracking site available? I already have my own SVN via dreamhost but am now thinking about use of a specific defect tracking system for defects/enhancements to track them. Note - I''ve gone for Dreamhost hosting. Not sure if there is a specific

Hi there I''m in the process of developing an e-commerce Rails app but am getting a little stuck on what models I should be working with on the order/checkout side of things. The app requires users to be registered and authenticated to checkout. So I already have a User model and an Order model (which belongs to a User). The Order model is largely similar to that used in the Agile Rails

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>