similar to: need BT102 firmware (current version)

FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC -> HP Switch -> Asterisk : actual throughput measured at 94.1 mbps. PC -> BT102 -> HP Switch -> Asterisk : actual measured at 8.86 mbps. The

Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA Bruce Komito High Sierra Networks, Inc.

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

Just got my (10) BT102 phones, flashed them to 1.0.5.20 and all work. No duds at all. Not a bad little phone at all. Doug

Just added a new Grandstream BT102 to my network. Its running new firmware (Ver 1.0.5.22 of 2005-01-21). I could NOT get the damn thing to (SIP) register.... Gripe 1: The New Firmware does NOT show the current version of all the firmware. You have to ask the phone manually with its menu button. Gripe 2: It does not show '****' in the the two password fields... This is what caught me - I

On Thursday 16 Jun 2005 09:25, Mark Brown wrote: > Hi Everyone, > > I'm using Asterisk, actually A@H 1.1 with all Grandstream 102 phones. > NAT is not an issue as all including the server have public IP's > > The problem is that the phones keep losing registration with the server. > I have not timed this exactly to see if they drop off with exactly the > same

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

Hi all, I'm quite new on this mailing list, and I discover the asterisk world. I m experimenting a PBX with SIP phones, grandstream budgetone (not expensive for tests) All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Here is my config : 2 sip phones BT102 with

Voicemail works fine for me. Post console output here to let us know what went wrong. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Breeden Sent: Friday, March 25, 2005 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: CAUTION: Re: [Asterisk-Users] grandstream

I have an * server running 1.0.9 on a FC3 machine. I connect around 44 BT102 phones to it and 6 Sipura 2000 units. Everything is working great but lately I have seen the following error: Aug 4 13:33:24 WARNING[31907]: chan_sip.c:4826 check_auth: Stale nonce received from '<sip:4000@148.235.174.85>' Aug 4 13:33:24 WARNING[31907]: chan_sip.c:4826 check_auth: Stale nonce received

Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/ Or directly from Grandstream at http://www.grandstream.com/BETATEST/Release-b21p1.0.5.23.zip Release notes doc here http://www.grandstream.com/BETATEST/Release_Note_1.0.5.23.doc while on the matter I just want to extend a note of thanks to Grandstream, I had 2 early handsets of theirs fail recently (about 9 months old)

Grandstream owners, I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ Doug

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've searched for a few days now without finding an answer. The release notes for version 1.0.5.16 of the Grandstream firmware says it supports attended transfer using replace but the docs haven't been updated so I can't work out how to enable it, or whether it should Just Work. I'm currently using the # attended transfer patch for * but would like to get back to using the

I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11. It's relatively stable, and the last thing I want

FYI to all you Grandstream users out there. I just fetched and installed the 1.0.5.0 firmware, and it appears they have removed the option to either do or not do SIP registration. Now it appears that one is going to register with the server specified in the "SIP Server" field, without any ability to disable it. Bug or feature? Just thought I'd put this out there and see if

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

>From the usual place, http://www.grandstream.com/BETATEST/GXP2000/ Note, there are two (and it took me a bit of a while to figure out) images to be loaded. Copy the ...a.bin's and the .bin's to your http provisioning directory, and reboot. The phone _must_ load the .bin files before it understands the ..a.bin files. After it loads the first one, the phone does lock up with the

Looks like Mozilla forgives that problem you are talking about too.. I have had no problems configuring the GS phones using Mozilla on RH9.. > -=> On Sun, 24 Aug 2003 09:47:06 -0400 (EDT), John Vozza <john@netrom.com> said: > > > Just noticed that version 1.0.3.81 has been released on the > > Grandstream website. > > Doh. I see they still haven't decided