similar to: Doorphone vs. Grandstream BT101

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

Hello, Before throwing in the towel with my Sipura 3000 has anyone had much success with that adapter connected to a door phone? In our setup a doorphone is connected to the SPA's fxs port. When a visitor rings, asterisk calls a group of Polycoms and the person who answers has to enter *1 to trigger the door opening. However it seems the SPA doesn't relay the DTMF's to the

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

Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of

Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart

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.

Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with

Hi, could anybody suggest a usable doorphone and magnetic door opener "hardphone" system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. thank you, Csaba

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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--1.0.0.6 Custom Ring Tone:

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

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 intercom purposes. Anyone have this working?

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

I have used the handytone 488 from grandstream in the past.... However I need to be able to send a number to a unit like the 488 and have it dial out. Is there a unit like this available? Basically a 488 unit that can place a call out. Jerry

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

Very friggen cool, that you very much for the information it looks like it will do the job nicely. What did you use in your extensions list to activate the relay? James > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jay Milk > Sent: Sunday, 20 February 2005 6:24 PM > To: 'Asterisk

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

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 Cotton

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 an external IP for the phone )

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the