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Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in

I'm now up&running with - mISDN with avmfritz driver for Fritz PCI card - chan_capi from debian recompiled with a patch (see below) - EuroISDN with Point-to-Point (ptp) mode (Austria) - With Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k from debian sarge But am having some problems: 1) I needed to patch chan_capi.c from debian sarge (see below) to give the new channel to asterisk in state

I've got a bit trouble with callerid and zaphfc cards. Basically zaphfc doesn't add the "0" in front of national numbers (haven't tried a international call yet). With chan_capi that allways worked fine, however i had to define the national and international prefixes in capi.conf. Is there something similar in zapata.conf ? Here is my zapata.conf: [channels]

I tried to setup an PBX System with the following parties: 1. SIP Phones 2. One AVM ISDN Card with CHAN_CAPI for Outbound dialing and receiving external calls 3. One HFC-S PCI Card in NT-Mode with ZapHfc for physical ISDN Phones. 1+2 work perfect I have problems with part 3 Card is working, drivers loading fine, asterisk initializes chan_zap without problems. Demo did work. My problem is

Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ? And in Italy, I often have set pridialplan = unknown About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with snom 190; with pa168s and ywh10 I have again some problem, the echo come up also after 1 minute of

hello! I'd like to use my asterisk-box on a p2p-line from my alcatel pbx. isdn4linux and capi works fine on a s0-line from the pbx, but i need more then 8 msns, cause i'd like to implement a voice-mail system and an extension to our "old" pbx with voip-phones. had anyone success using isdn4linux on a t0 (p2p) line (of course with an inexpensive passive card like my avm

Hello, I've added a ZAPHFC card to my CAPI based system. Calls coming in via ZAPHFC do not forward the caller id to the SIP phones. Calls coming in via CAPI do forward the caller id to the SIP phones. Any and all help is greatly appreciated. The (hopefully relevant) conf file excerpts are: extensions.conf =============== exten => 807440,1,Answer exten => 807440,2,Noop exten =>

Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed

hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another

Hi all, I really hope that you guys can help, because I've been tearing my hair out for the past 5 hours on this one. I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel Meridian phone system. Phone calls from the Nortel to say MSN 510 are correctly being sent to the right SIP phone. When asterisk dials say Zap/g2/224 (a Nortel internal extension) the call goes

Hi, I've been learning asterisk for a couple of weeks now - and it worked for me as faar as standard configurations where concerned (sip/iax outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO). Now I recently I'evaluating to use asterisk as a replacemnt for our companies (15 employees) legacy pbx system and I'm experiencing multiple problems with the hfc isdn cards:

wondering if anybody knows this......does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. Nauman -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, July 08, 2004 5:48 PM To: asterisk-users@lists.digium.com Subject:

this happened to me on a cvs update, rebuilt a clean chan capi cm and all is well. Greg ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Voicomm User Sent: Monday, September 19, 2005 3:29 AM To: Armin Schindler Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk ISDN: Problem

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. the customer is currently using a paging system (small receivers which display a callback number and a base station (transmitter) with several antennas at the site) the problem is, that the currently operative base station uses 4 ISDN BRI interfaces. But the protocol is old germany 1TR6

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten

Hi- Anyone using the Junghann's quad BRI card and the Digium E100P in the same system? I'm having a configuration problem where I can configure the cards one at a time (with the appropriate drivers loaded) in a system, but when I try them both together, neither will work. They both work fine one at a time. Probably has something to do with the channel numbering. I've tried

Hello, I need to detect both fax and answer machine, and it should be valuable that the detection will be run by the Diva card itself. So : - I read Diva Documentation, and I found that the Diva could send some specific DTMF, if I had "[..] enabled [this functionnality] by the application for a designated controller through a manufacturer request command 9 [...]", but I didn't

Hi list, I am searching for a possibility to do a certain call transfer method which is called "path replacement" in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and

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

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work