similar to: BT101 and caller id and web interface

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'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

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?

Hello all, This email is intended rather informative than questioning. While developing some script-generated dial plan, we figured out that there are differences between Snom 190's and BudgeTone 101's relating to transfers. It appeared that the 190's will have their own 'Caller ID' set as the 'CALLERID' variable in astersisk when transfering a call, while 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

Hi, We have to servers set up in two different networks. We are able to connect calls via IAX and they work perfectly. We do not see caller ID from clients on either side. Our Grandstream phones say Eri and our XTen phones say Asterisk. We did a debug and I am pasting the output from both servers below. We tried setCallerId in several different ways. We see the value get passed to the

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

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 tool on the GS website to generate

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

Hi, When we use BudgeTone-100 in our Intranet together with our Asterisk IP PBX everything is working OK. When we try to use the phone behind the Firewall we can't do the connection. When I try to use STUN Server: 128.107.250.38 there is no result. The only way in which I have audio from the one direction (BT-100 to Asterisk) is when I leave blank STUN Server and specify the IP Address in

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

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

I can't seem to hide and/or set my caller ID from *. I'm using a quite recent (three weeks or so) CVS with an E400P card. I have pridialplan=unknown in zapata.conf and I'm based in the UK. The relevant bit of pri debug looks like this (reformatted to fit 80 char width): > Calling Number (len= 4) [ Ext: 0 > TON: Unknown Number Type (0) >

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

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 got a broadvoice "business" account under the byod(bring your own device) program. I have applied the patches and created new asterisk debian packages. I have the account working on inbound and outbound. The problem area is outbound caller ID. I have 3 other accounts with IAX2 providers and have no problem setting the caller ID on outbound calls. I called them and they

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hi, I have been looking for an example of accomplishing this, but I've been unable to locate something similar to what I'm trying to do. Here's the scenario: Users caller ID is set to their internal extension (200-250). This is set in sip.conf for each user. Each user has a local DID as well (hosted through Vitelity, for example (555)111-2222). The problem is that this extension was

Nicol?s Gudi?o <asternic@gmail.com> wrote: >>> Paul, 1.0.5 stable suffers from caller id issues as well, at least for >>> SIP channels. What fixed things for me was swapping in app_dial.c from >>> 1.0.2 stable (didn't try others). You could also just diff app_dial.c >>> between versions to find the problem but I took the lazy way out the >>>