similar to: Incorrect sound

Hi, Bit of a puzzle this one - let's see if anybody else can shed some light... I Ghosted my Asterisk box to build one for a colleague. I added one HFC card to make the total two and amended zaptel.conf and zapata.conf accordingly. I tested it with my handsets on my ISDN lines and it worked AOK. I unplugged the box and took it to his house and plugged it in. He doesn't have any ISDN

Hi all, I've been batting my head against a brick wall for the best part of the day and still haven't got any further (apart from getting a big headache, that is). I've searched the Wiki and Googled the hours away, but I still can't find supportive documentation. I've just replaced my ISDN Fritz!/chan_capi setup with a HFC/Zap configuration and had the following problems: 1

Salve, I'm somewhat stuck on how to get DTMF working with my setup and googling didn't yield anything similar. My setup consists of one CAPI-capable board (AVM Fritz!DSL) connected to a BRI (T-ISDN), one HFC-S board running in NT-mode connected to an internal S0 bus with some ISDN devices (DECT stations, TA) and, of course, some ethernet interfaces. ISDN standard used is Euro-ISDN.

Hello, I am trying to use a HFC-PCI (CCD/Billion/Asuscom 2BD0) card in TE mode to dial-in and out with ISDN. The problem is I can not get the card to dial out with a Zap channel. Dial-in is working. I am using bri-stuff 0.1.0-RC4 (but tried also RC3 and RC2k). I tried all combination of "immediate", "overlapdial", "pridialplan". I earlier also managed to dial out

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

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

Hi all, Maybe I'm being thick here, but I've had a look through the mailing list and the Wiki, and I can't seem to see details of anybody else with this problem.... With the following line: exten => s,1,HasNewVoicemail(201) I am getting the following error: -- Executing HasNewVoicemail("SIP/201-2f1e", "201") in new stack Sep 2 12:41:09

Hi, I would appreciate some advice on the following: how does one use a single BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and Asterisk 1.8? For Asterisk 1.4 I would first install mISDN & mISDNuser and then build Asterisk 1.4. Now I noticed that Digium's HFC chipset based B410P is supported by the latest DAHDI & libpri but reading the source of the

On Friday, March 31, 2006 3:52 PM Chris Earle wrote: > I'm wondering if I should be using zapHFC with my Junghanns card > instead of qozap? Why would you want to do that? Sorry if I missed the start of the problem but qozap is what you want to use with your Junghanns card (at least if you want to stay with zaptel/libpri). > Everyone always mentions zaphfc -- mostly I >

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

How can I get external (telecom local exchange) dialtone on HFC ISDN BRI with bristuff/zaphfc driver? with capi, voip-info say that it should be something like: Dial(CAPI/MSN:b) But with zaphfc, if I try: Dial(ZAP/1/), I just get NOANSWER.

On Saturday, April 15, 2006 3:17 PM Remco Barende wrote: > I heard that Junghanns is working on such an interconnection. It is > already possible to connect their PRI cards, and they are working on > BRI<->PRI. Correct. The next driver generation is supposed to support this fully. > I ise their bristuff for an HFC-S BRI card and am not happy at all > with the way they

I have a question, I have to provide a solution for an office that will be almost abandoned, and there will be one or sometimes two persons 2 days a week. The main number however should be preserved. They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20

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, Bristuff works great with HFC card... your compilation problem may come from your kernel configuration... You should check this doc, at least, for the redhat config : http://www.automated.it/guidetoasterisk.htm Then, installation of Bristuff works like as charm ! Bye David Masure -----Message d'origine----- De : vdasilva [mailto:aquamoon@wol.co.za] Envoy? : mardi 28 juin 2005 09:01

Hi Is anyone having this weird trouble with bristuff....calls don't get answered, but on the debug it looks as if the call is answered. The call will just hear ringing. I am using -BRI-stuffed-0.1.0-RC4a Occassionally the system starts working on its own, other times I have to stop and rmmod everything and start them all up again. These error messages appear on the CLI: == Primary

I have grouped two capi controllers using * as the outgoing msn. when the first two channels are busy normally and I try to use a third channel the channels from the second controller are used. But when a channel is occupied by the capisuite fax and we need a third channel asterisk responds with "every one is busy congested at this time". I think the same behavious would have happened

Hi Manfred, I applied the patch and recompiled and reinstalled and I got the folowing warning during my first test call: Aug 19 12:26:51 WARNING[294927]: dsp.c:1234 __ast_dsp_silence: zero length packet It looks like that could be the problem... and the fix! I'll let you know if the problem reoccurs. Might it be an idea to submit the patch to the bugtracker? Thanks, Gary ----- Original