similar to: Re: time to build an open phone?

ACES - Asterisk Communications Endpoint System {the following could be used by any IP-PBX but the name pays homage to Mark Spencer and friends who cannot be lauded enough for their fine work} As you read this it will be obvious I am not a professional engineer but I do have enough knowledge to be fairly certain what I'm proposing is feasible from not only an engineering, but production

Hi all I have a very strange problem here... I use a hfc-s card with mISDN in NT mode with an ISDN Phone connected. When I make a call, the phone rings two or three times and then misdn runs into a timeout... I don't know where to set that timeout, but it's way to short for the called to pick up the phone. If the destination phone is picked up, then everything is allright and the

I'm running a setup with chan_misdn on a austrian PTP-line. When somebody dials in without an extension, he gets a busy signal. I don't see the call at all in asterisk. I *have* set immediate=yes in misdn.conf. And I *do* have an s-extension in my dialplan for the context used by misdn. Calls that provide an extension work fine. Attached is my misdn.conf and a verbose 3, misdn set debug

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

Hi I have beronet BN8S0 isdn card in my asterisk and , card is working fine, but when I try to dial to special number 118913 ( telephone number information) from polish telecom TPSA, I always geting timeout . Bellow is isdn signaling dump : --> * CallGrp: PickupGrp: --> rxgain:0 txgain:0 --> * dad:118913 tech:mISDN/2-u25 ctx:default --> * Setting Context to from-tpnet -->

So has anybody got one of these? http://www.amazon.com/exec/obidos/ASIN/B0007LQQUK/qid%3D1106972010/sr%3D11-1/ref%3Dsr%5F11%5F1/102-1529886-6420131 I'm thinking that it should be possible to connect it directly to an Asterisk box and not use their software, as long as there was Linux support for the USB dongle. Maybe it just looks like a standard USB audio device? But there has to be an

hi i've configured a Beronet BN8S0 Card with misdn beronet utility. the card is configured with all lines in TE and P2P mode, and it is connected with the special cable with an ISDN connection. i've turned on debugging to level 4, this is the output from the asterisk cli when i receive a call: P[ 5] MGMT: Short status dinfo 1000001 P[ 5] MGMT: SSTATUS: L1_ACTIVATED P[ 5] handle_frm:

Hi. I'm running Asterisk server with 10 sip phones, and 2 grouped T0 lines with 10 DDI numbers. My provider is France Telecom and my setup is : - Debian Lenny - Asterisk 1.4 - Linux kernel 2.6.25.17 - mISDN 1.1.8 driver - Sip phones Thomson ST2030 No problem with the SIP . But when reveiving a call on RNIS line (any of the DDI numbers), the associated SIP phone rings indicating _two_

interesting for anyone testing out speex :) kfish. ----- Forwarded message from Alan Clark <alan.d.clark@telchemy.com> ----- From: Alan Clark <alan.d.clark@telchemy.com> To: avt@ietf.org Date: Mon, 13 Sep 2004 12:57:01 -0400 X-Mailer: Microsoft Outlook IMO, Build 9.0.6604 (9.0.2911.0) Subject: [AVT] Open Speech Repository We've started to build a database of speech samples in

i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. My extension conf is: general] static=yes writeprotect=no [globals] OUT_PORT=1 [ISDN] exten => 12345,1,Dial(SIP/012346737222 at sipprovider.local) If i call to the msn 12345, the SIP-call is going out, but after

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now

I'm trying to set caller ids on outgoing calls. I have a quad BRI B410P card connected to my telephony provider. I know the list of DID numbers the provider assigned to my company. If I don't set the caller id then the callee always sees the same "top-level" number. If I set the caller id to a specific DID number we own, the callee keeps seeing the "top-level" number,

Hi, Since a week ago I am trying to get chan_misdn working with asterisk 1.4.19, using HFC based ISDN card on Linux 2.6.22. My setup is done as detailed on wiki and FAQ. * mISDN and miSDNusers are 1.1.7.2, unpacked, build and installed. After installation and misdn-init, I have this: aragorn:root/pts/1: # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) ->

Use Set(CALLERID(num)=9999999999) instead of using CALLERID(all). Remember to set this BEFORE you Dial. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 12:36 -->> To: asterisk-users at lists.digium.com -->>

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

You're quite right. We'll need to see your misdn.conf file to check the settings. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 13:49 -->> To: asterisk-users at lists.digium.com -->> Subject:

Some smartcard readers have keypad to enter the PIN securely (i.e. such that it cannot be intercepted by a rogue (ssh) binary. PKCS#11 allows for enforcing this in hardware. Below patch allows for SSH to make use of this; against head/master as of today. Dw. commit 7f0250a8ae6c639a19d4e1e24fc112d5e2e1249a Author: Dirk-Willem van Gulik <dirkx at webweaving.org> Date: Tue Mar 17

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Hi. I want to handle incoming chan_misdn traffic by asterisk, but I've got message - 'Extension can never match, so disconnecting'. What I'm doing wrong ? How I can pass incoming dialed number (dad) to misdn context (in my case 'dss1_incoming') ? Works unrouted calls (s extension) if I set immediate=yes in misdn.conf, but I want to route calls by dialed number. log