similar to: zaphfc NT-mode can't dial outgoing

Sorry that I wrote in german : Ich benutze asterisk mit dem zaphfc Treiber. Jetzt hab ich folgendes Problem, habe 2 ISDN-Telefone angeschlossen. zaphfc im nt-mode. Anrufe von ausserhalb per sip (sipgate.de) kommen an. Wenn ich aber intern zwischen den zwei Telefonen (Ascom Eurit 30) sprechen m?chte geht das nur wie folgt : Erst die Nebenstelle w?hlen und dann den H?rer am Telefon abnehmen.

Hi list! I am installing an * box that will be installed on a site with KPN BRI ISDN in The Netherlands. I am using bristuff fron Junghanns. Does anybody know the correct settings for this? I will not have internet access there which makes it harder to google around on location. switchtype = euroisdn is pretty obvious but what about these settings: signalling = bri_cpe_ptmp ; p2p TE mode

Hi I have a small ISDN PABX , (BRI) at home - Siemens Gigaset 4175 which has cordless DECT extensions. I have set up asterisk on FC3 with two HFC cards and I am using the latest bristuff. I am trying to use * between my ISDN line and my PABX. One card is in TE mode and can receive and make calls OK. I can make and receive external calls to a SIP phone. I cannot get the other card to talk to my

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]

Hi, I have a partial working installation with zaphfc. Incoming call : For incoming call, seems work fine. But the sound is very bad with bounce short crashing sound. Same sound with echo cancel off or on. SDA work fine. Another problem, it's seems that's zaphfc don't reset correctly the line. I have one of my D channel how was busy even after stop communication. Outgoing call :

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

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

I have some problems starting asterisk with a hfc card using zaphfc: [chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading module chan_zap.so failed! I'm using asterisk from debian/sid: Asterisk

Hi, i've downloaded/compiled/installed the bristuffed asterisk Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine with kernel 2.6.11. Asterisk works well if i configure the card using isdn4linux. I'm having problems dialing out (not tried the input yet). This is the output from asterisk: -- Accepting AUTHENTICATED call from

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

Hi Guys, - if i buy the codec online NOW, will this license work with the new one, or shoud i wait 'til it is no longer beta? - how does the registration work? Will there be a key-file that i can backup and use on a new installation of the same machine with the same hardware, or do i have to re-register on any new installation (and therefore i'm screwed after the second new

hello, I have a asterisk box (Slackware 9.1.0, Linux 2.4.31) connected to a Ericsson Businessphone PBX on the internal S0 bus with HFC-S card and zaphfc driver - point2point mode -. --------- | TELCO | | BRI | --------- | | PBX external S0 -------- | PBX | -------- | PBX internal P2P S0 NT Mode | | HFC-S Card P2P TE Mode

Ok, now thats a gadget i want to have :-) http://www.global-teck.com/english/newproduct.php http://www.global-teck.com/english/telecomproducts.php Anyone knows something similar that would work with asterisk, or any chances getting this to work? Regards, Andreas _________________________________________________________________ Need more speed? Get Xtra Broadband @

A little off-topic: I have the following hardware: 2610 XM NM-2V VIC-2BRI NT/TE IOS loaded: flash:c2600-ipvoice-mz.123-5d.bin" I get the following error while booting: %C542-1-UNKNOWN_VIC: VNM(1), vic daughter card has an unknown id of FF Is the VIC-2BRI compatible with the 2610XM? What IOS needs to be loaded? http://www.cisco.com/en/US/products/hw/modules/ps2641/products_tech_note0918

I have an asterisk box setup with a T1 card (hooked to a pri from the telco). Sometimes when people call a number that rings down that pri, the first ring is really really long, like 3 normal rings put together. My indications.conf is the default that comes out after you do a make install. Is this (first long ring) something I can control?

Hello all, I have a problem with overlap dialing and don't know how to get rid of it. My setup is: 1 HFC card with bristuff -> ZAP/g1 (2B + 1D channels), SIP phones (I just removed TDM400P with 4 FXS) I created test extension 222 which goes directly to g1. In Zapata.conf overlapdial is set to yes. First I created this extension: exten => 222,1,Dial(zap/g1,100,tc)

Hi! Has anybody seen anything like this using zaphfc? On outgoing calls (via isdn) , the line gets hung-up as soon as the called party answers. As seen below i get some protocol error (6) - but i'm not sure if this is related to the "hang-up" which apparently comes a little earlier?! Incomming calls on the isdn (zaphfc) interface is working just fine (P.S. what about the

Try commenting out the line pritrustusercid = yes Or set it to 'no'. That worked for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jens Sent: Friday, January 21, 2005 7:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zaphfc no callerid

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 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