similar to: Best Grandstream firmware to use?

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'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32 C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25 Jan 18:25 Error Aborted 192.168.16.32 C:\Program Files\TFTP

Hi all, Just wondering if anyone's come across this issue, and what might be a fix for it: We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The phone can do proper supervised transfer, but _only_ once. If the user attempts to transfer a second time, it won't work. any suggestions/hints/tips are welcome.. Flynn

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

>I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN?

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

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

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

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

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]:

Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David

Hi, I'm confused as anything by this bug. I'm hoping that it is just something screwy in my config. I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium TDM31B. I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both Asterisk and the Zaptel driver on Fedora Cora 1. When I make an outgoing call on the Cisco phone, everything works fine. I'm

I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack

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.

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

hi all i upgrade a bt100 phone and it can't resgister with asterisk Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from '<sip:@172.16.4.249>' failed for '172.16.4.226' is was working with the version 1.0.5.3 some bady now what is hapening? thanks in advance Rodney

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

Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometimes it doesn't, etc. I've come to the same conclusion as the wiki: it's probably better to avoid this

Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The

I'm getting 1.0.4.30 I think it is, in new phones, but all that's on the website is 1.0.3.81 Where do you download newer versions ? And, will anyone else's firmware work on these ? This firmware seems to be flaky at best. These Budgetone phones SUCK with NAT involved.