similar to: Grandstream BT102

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

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.

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

Would anyone have a copy of the latest firmware release for the grandstream BT102 phone? seems grandstream no longer offers it on their website (of if I missed something a link would be much appreciated.) Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090504/9ba2730e/attachment.htm

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 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

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

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

Is there a a consensus on which of these is the better phone.I've personally been using an 841 and have learned to live with its shortcomings. I now need to recommend some phones for some sites we're installing. I'm looking at the BT102 for desktops that don't want/need a headset but need a phone for the higher end users (without costing the earth). TIA, tony Zero Effort

can anybody identify why the CLI is issuing a failure message while debug shows everything is fine???? this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]:

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

Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug

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,

Hi there, what phones are available that have two ethernet ports? I want to do some cabling at a new installation and i heard there are such phones (SIP i guess) out there. That way i dont have to run two cat5 to the user desktop. I think 3COM had one but can't find the web site reference for the two port phone thanks, erick

I have an nec electra elite 192 with a t1 card; and am looking for suggestions as to integrating them (can't throw out the system yet!). I have a fully working asterisk server -CVS-04/27/04-19:01:05- (found a hp d220 for $350.00!), 2 digium t100p cards, a plain t1 with loopstart signaling, and 2 working bt102 grandstream ip phones (thanks again Matt for your "start from scratch"

Just to toss in my very limited experiences with the Grandstream phone-- I haven't tested it enough to really know nor is my Asterisk config set up enough to fully try all the features. Mostly, it just works. It was very easy to configure and get running. I've been toting it around to clients as a show and tell exhibit and it has helped get people excited about the possibilities. Voice

On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > Doug, > > I tried that as well. Even with my dialplan looking like this: > > > > Ordering by includes works for me under Asterisk 11 and 13 > > What does the output of the below show? > > dialplan show from-external > >

Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul.

> exten => _13XXXXXXX,1,dial(${OPERATOR},20) Hello "SIP/2.0 401 Unauthorized" Unfortunately the negative. An asterisk indicates a 404 error. On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle <support at drdos.info> wrote: > > On 3/5/19 2:46 AM, Gokan Atmaca wrote: > > Asterisk can send calls, but I don't get a call. What could be the problem? > > >