similar to: New to the list.

Hi Asterisk Community, I have a development with a TDM11B in it. I am trying to connect this to a exchange line as well as a UK telephone and are looking for some pinout information for the FXS port. Is it the centre 2 pins that at the tip and ring. I have been digging around the Digium site but cannot seem to pick up this info. Any help appreciated. Thanks -- Gary

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Hello list, I'm observing what I believe to be inconsistent behaviour regarding "Newstate" AMI events for the "Ringing" state. As such I come to you asking for experience or advice: am I wrong or should I file a bug ? I present you a short introduction which I feel is relevant; however, if you want to go straight to my technical question, please scroll

I have a SPA-2001 and I didn't realize I could use calling features on an analog handset. Does that mean you can dial *77 and use a VOIP feature? (like forward or hold)? Mike ________________________________ From: Jorge Carrasquillo [mailto:jorge.carrasquillo@gmail.com] Sent: Thursday, June 23, 2005 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea?

After two weeks of reading about asterisk and joining this mailing list, I finally decided jumping on the asterisk bandwagon... Asterisk rocks!!! I have a www.Stanaphone.com SIP (free) for incoming line and a www.VOIPJET.com IAX line for outbound. I also have a www.Vonage.com line (gives me 500 outbound minutes) and a Cingular cell phone (gives me 800 minutes) and I also use Skype fairly

<div><BR>Hello,</div> <div>&nbsp;I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same question....how do I go

See: http://lists.digium.com/pipermail/asterisk-users/2004-September/063348.h tml -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kumara Jayaweera Sent: Tuesday, June 14, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease

Hi, I got my TDM11B and am trying to get it to work on my PC. However, I'm having difficulty getting the wcfxs driver to load. I've Googled this problem, and while there are others who have ran into the same problem, none of the solutions work for me. I would very much appreciate it if you could suggest things I could try to resolve this problem. Details follow. The PC is somewhat

Hey, all. I've got a client who's interested in possibly using a softphone for his receptionists. While I've certainly used some softphones for single extensions, I'm not sure which one I'd suggest for a receptionist. Any favorites? Thanks, -Ken

The long waited Ultimate GSM Gateway is finally out. This time we have managed to source a new patch of brand NEW GSM Gateway at prices that is only 50% of what the market rate. And with the SMS Function and many more... For purchase please email gsm AT cyper-telecom.net. We accept paypal and bank transfer. Postage is not included. Please notice we have also got the standard Dual Band GSM

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

Sauda??es. Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o conector RJ45 est? crimpada com a sequencia 568B e vai ser conectada na placa Khomp, mas a ponta RJ11 eu n?o sei como deve ficar. Li alguns manuais na internet mas n?o entendi ao certo como tem que ser feito. -- Att.* *** Luis H. Forchesatto

I've got an interesting scenario where transfers while getting an invite seem to break. Here is the scenario: You have a receptionist who has a 6 line phone (in this case, a polycom ip600 - also tested with a Cisco 7960) the receptionist has all six lines available for use (in the case of the cisco I tried registering all lines as one number as well as registering multiple lines and

Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the

I have a customer who wants their receptionist to input the users' long distance PINs for the because they use each others pins. I am having trouble coming up with a way to do this because of creating a channel between the user and receptionist, dropping the channel and its variables and creating a new one for the actual long distance call. Any advice is really needed. 1. User Dials Long

I get this error no matter what I do. I've tried switching to different locations on the TDM400p to no avail. I've got the card setup with the green module in slot1 and red in slot2. My config files are exactly like the "Configure Dev-Kit PCI (TDM11B)" example. http://www.digium.com/index.php?menu=configuration#TDM11B When I run ztcfg -v, I get: 2 channels configured.

Hi! I wish you all e Happy New Year first! Allthough, I'm relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. :) Platform is Debian 8/Asterisk Packages (11) from Debian Repo. But I am running into problems setting up 2 older Hardphones, Thomson 2030S. :( with in my sip.conf, I have got for this hardphone: [...]

Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: "user A" <----- pstn -------> ASTERISK <----- mgcp ------> "user B" "User A", connected behind a PSTN, tries to call "User B". After dialing "User B"'s number, call comes to ASTERISK, ASTERISK contacts

hi there, I saw a page on voip-info about the thomson ST2030 phone. There is not so much info on there, that's why I would like to raise a question here. Has anyone got hands-on experience with this phone? (with or without extension module) I am interested if it can be used (as SIP phone) in a good way with asterisk. (also, do all the functions behave like they should; like Supervision