search for: bt101

Got one weird one and one prob easy one. 1. I have upgraded our BT101's to Program--1.0.5.0 Bootloader--1.0.0.17 HTML--1.0.0.34 VOC--1.0.0.6 after doing this i have some phones on different subnet's ie 255.255.255.248 or .192 or .252 and i am now unable to login to these phones from different subnet's . I have one at home which is on a .248 ( Using...

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 on-hook,...

Hello, Has anyone noticed that if you don't pick up a BT101 phone in 60 seconds it stops ringing and acts like it was never called ? Or is it just something I missed ? If it matters for something I have call waiting enabled on the phone. Product Model: BT100 Software Version: Program--1.0.5.11 Bootloader--1.0.0.18 HTML--1.0.0.37 VOC--1.0....

Hi, I am wondering if anyone has gotten the BT101's to work with the paging in Asterisk? I know that the phones themselves have an auto-answer option and if I turn it on every call is auto answered. I want to be able to call the extension normally and have it ring normally but if someone dials # and the extension to have it auto answer for i...

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

I know this must have been asked before, but I was just wondering, the manual says it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

Hello, We found out that after upgrading the firmware in our GrandStream BudgeTone phones, that we were not able to transfer calls anymore. We use the BT's own tranfering mechanisme. We can dial the phone where the call should be tranfered to. But after that, the original caller stays in music on hold on the server and there's no way to get the calling channel back again (not to the

...loader into the asterisk's CALLERID variable). I've found some references that said that the differences between the 190's and 101's might be the result of unclear definitions the the SIP transfer RFC. Anyone figured out how I might get a transfering party's caller ID on a BT101? Otherwise I will have to trash the 101's (I hate these garbage phones anyway, but it would be nice for customers if they're able to choose between 101's and more expensive 190's). Using asterisk 1.0.7 + bristuff RC8g; version 3.56m on the snom190's and version 1.0.6.3 on th...

....g. from iaxcomm) gives the following on the console: -- Accepting AUTHENTICATED call from 192.168.112.99, requested format = 512, actual format = 512 -- Called 0031651931985@mutualphone -- SIP/mutualphone-6b26 is ringing -- SIP/mutualphone-6b26 answered IAX2/iaxrene@iaxrene/2 The BT101 gives this: -- Called 003165193XXXX@mutualphone -- SIP/mutualphone-2de1 is ringing -- SIP/mutualphone-2de1 answered SIP/chimit01-6013 -- Attempting native bridge of SIP/chimit01-6013 and SIP/mutualphone-2de1 Jan 16 18:50:41 WARNING[18631600]: chan_sip.c:2804 process_sdp: No compati...

I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a...

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 -- Exe...

...ot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls. After a reboot is works fine again. We have a * box with many BT101 and softphones working for months without any problem. I'm missing something? it is a bad config file? or it is a phone bug? Thank You for your time. Jorge Mendoza

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 to do...

...problem on my LAN only. I replaced the LAN switch with a new linksys 2024 with QOS and seemed to help but not fix the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are cheap that are known for echo problem in the handset. I have one remote user that never has a problem. I have a remote test server at home connect via IAX with no problems, also a PAP2 with no problem. External faxing from the rest of the world via our voip provider is...

...hen a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-December/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html )...

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

Up until yesterday I've had a lot of high pitched noise when connecting a BT101 to the PSTN via the X100P. I was using an Asus A7V133 with raid motherboard and an 850 AMD Duron. Over the weekend I thought I'd try another machine. I had an HP Vectra 400 Mhz PII MMX with 128 Mb RAM available, today no noise at all. Now I must see if the Vectra is up to the job. -- Dave...

Some CVS upgrade in the last day or two has broken the recognition of DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting the error... *CLI> -- Executing VoiceMailMain("SIP/phone1-e0dd", "") in new stack -- Playing 'vm-login' (language 'en') **Here I push a button** May 9 18:26:18 WARNING[98311]: chan_sip.c:5027

.. and from a newbie no less :-) I have configured my BT101, and hooked it up to my * box. All is well. I have entered the following in externsions.conf, and this bit works: exten => 613,1,Answer exten => 613,2,Playback(demo-echotest) exten => 613,3,Echo exten => 613,4,Hangup If I pick up the BT101, and dial 613, sure enough I get the echo t...

I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in the sip.conf). Doe...