similar to: Writing messages to ISDN phone displays?

Since the largest regional ILEC in Brazil completely stopped taking new BRI ISDN customers and started heavily incentiving their existing customers to migrate over to ADSL (regardless if what they need is voice or data), we have thousands of ISDN phones collecting dust. Is there any network side U cards that will work with Asterisk, so we can turn those phones into useful things ? I'm

Hi, I finally got Asterisk installed using the German installation CD from http://www.asterisk.de.ms. I got two SIP phones working (SIPPS) asterisk*CLI> sip show inuse Username incoming Limit outgoing Limit 5678 0 N/A 0 N/A 1234 0 N/A 0 N/A

Hello, I have a Gigaset S44 connected to a quadBRI NT port. Receiving calls works phone, however when dialing out from the phone the call is dropped to the 's' extension, as if no extension had been dialed: -- Accepting voice call from '492389990' to 's' on channel 0/2, span 4 -- Executing Directory("Zap/11-1", "default") in new stack --

I've a strange problem with CallerID when calling Vodafone mobile's from my Asterisk Box. If I dial out on my ISDN-30, setting the CallerID to my DDI (XXX802), a Vodafone mobile correctly shows the incoming call with this number and if the phone's not answered it shows a missed call from XXX802. All very good. However, if the call is not answered and goes to the Vodafone messaging

Deutsche ?bersetzung folgt / German version following ===================================================== Hello, i have Asterisk running with 2 ISDN-Cards. One AVM Fritz for connection to german ISDN and one HFC-compatible-Card (NT mode) for connection to ISDN-Phone (later: ISDN-PBX). Here is my actual installation: ISDN -> Fritz - ASTERISK ? HFC-NT <- ISDN-Telephone If i pick up my

Hello all, I am having trouble getting gtalk to work with my account which is not using a gmail.com email address. When I do this there an error from the Jabber module: [Feb 3 20:51:17] ERROR[6286]: res_jabber.c:573 aji_act_hook: JABBER: Node Error [Feb 3 20:51:17] WARNING[6286]: res_jabber.c:1495 aji_recv_loop: JABBER: Got hook event. JABBER: gtalk_account INCOMING:

Hello. I have problems to integrate an isdn phone. I don't know why but the isdn phone rings only once and than it looses its connection to his base station. I can make a call from the isdn phone to an VoIP Phone inside my network but when i pick up the phone the isdn phone also crashes. misdn.conf: [ntport1] ports=5 context=isdn-telefon msns=* extensions.conf: exten =>

Good day all, we have following setup: Debian Etch 64, Asterisk SVN-branch-1.4-r66244 with mISDN 1.1.3 and 2 Digium cards B410P. Our customers calls in the office through ISDN lines and then get a possibility to join meetme conference. It works well except when customers are using SIEMENS phones (not IP): DTMF is not recognized. Does someone have an idea on what could be the problem with

Folks, (sorry for overlong lines) I have recently configured one port on my QuadBRI card to work in NT mode with NET signalling configured so that I can use an ISDN telephone on it. I have set up a separate group in zapata.conf and can call the phone and place calls from it like a charm. No problems at all. Problems came up when trying to hold a call and get it back. I turned on "pri debug

Hi, I'd like to use Asterisk with ISDN interface and normal analog interface to door phone (or any other low cost connection type to door phone). What would be your recomendations for needed HW in Europe? Is it possible to have this in one PCI card? Are there any lower cost voip door phones? Thanks in advance, Robert.

Hello everyone, I'm using very simple setup to make and receive external ISDN calls through a softphone (x-lite version 3.0 - Win32) via an asterisk box. Hardware setup: - Dialogic Diva BRI (lspci yields: Network controller: Eicon Technology Corporation DIVA Server BRI-2M/-2F (rev 01)) - ISDN BRI line Software setup: - Redhat 9 - asterisk-1.4.16.2 - chan_capi-1.0.2 Asterisk configuration

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station

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

Hello all, New to the list, Asterisk and this wonderful world of convergence. From reading the docs, list and websites I hopefully correctly understand that I can use a plain cheap isdn4linux supported ISDN (passive) TA to hook up the PC to the ISDN line from the telco (FXO card?). Next in the PC I also need a FXS card like a Digium TDM10B or PhoneJack so I can hook up an analog phone to it. In

Hello everyone, I try to connect the Asterisk PBX Server on my NTBA (NT1) - actually its "hard"-connected on the box, but I've tried a crosslink-ISDN cable, too. My ISDN-Phone is connected on one port (of two) S0-Ports on NTBA. When I start Asterisk and want to make a call I'll geht a lot of this messages --> 040827-034639 WARNING[213005]: chan_zap.c:6902 zt_pri_error:

Hi, We're using Asterisk with Digium TE110P card for the PSTN E1 interface. Our PRI is enabled with detecting the "connected-party-number" feature. When an OUTBOUND call is made to a phone, the PRI will send back an ISDN messages containing the "connected-number" and we can use that information to validate the extension user is calling the party that he/she is

Hi, I'm looking forward to install an asterisk-server to perform ISDN-to-SIP- bridging. But for the times when I'm not available via SIP I also need call- forwarding-features. Afaik it is possible to directly forward a call on ISDN instead of opening a second ISDN-channel additional to the incoming-ISDN- channel and forward the call "by hand", right? What if I need features

Hi All, First of all, please no flames about the footer of this message. It's being inserted into all outgoing email by the server here without asking me personally first once I send out email. Since I've been very interested in setting up an Asterisk PBX + IVR + Voicemail system to support a small business and a foundation for quite some time now, in order to replace their current ISDN

Hi *, This is my first message to this list, so I hope I am not breaking any rule with my post. My question is the following: can I establish a voice connection over an ISDN interface by using a bluetooth dongle to connect to the ISDN access point? Here follow the details: I have compiled and configured asterisk. Everything seems fine except that when I configure asterisk to use the ISDN

Hi Folks, First of all, I've been reading a lot and a lot about asterisk and ISDN-BRI, but is really a complicated world, not like PRI or Analog. I have 2 offices with 2 NT boxes and I want to connect them with Asterisk to route outgoing from one office two the other. At the end I want them to look like this Office2 will have only VoIP phones (not ISDN phones connected to a card in asterisk)