similar to: DATA calls answered by IVR, but I don't want that

When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise that this is DATA call, but answering anyway (playing IVR messages, etc...) How to stop that? I want that only VOICE calls are answered, and DATA/FAX to be ignored. (I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, ZapHFC) Log: -- Accepting data call from 'XXXXXXXX' to '3001' on channel 0/2, span 1

When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise type of call, but answering anyway (playing IVR messages, ringing phones, etc...) How to stop that? I want that only VOICE calls are answered, and DATA/FAX to be ignored. (I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, ZapHFC) Log: -- Accepting data call from 'XXXXXXXX' to '3001' on channel 0/2,

When incoming DATA calls arrive on ISDN, Asterisk recognise that this is DATA call, but behaving like it is voice call: Answering call, playing IVR messages, etc... How to stop that? I want that only VOICE calls are answered by Asterisk, and DATA/FAX to be ignored. (I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f with ZapHFC ISDN BRI lines)

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

I'm not sure that this problem is strictly related to zaphfc, but this is what happens: my asterisk (build on bri-stuff-0.1.0-RC2k) handles a single PCI HFC-S based card. I own a Cisco 7940 Sip phone (fw 7.1) and a pc running X-Lite. Zaptel.conf and zapata.conf are taken directly from zaphfc samples. Extension.conf contains the following lines: [from-ISDN1] exten=>s,1,Wait(1)

Hi, I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel : Linux laps1 2.6.13-15.7-smp #1 SMP Tue Nov 29 14:32:29 UTC 2005 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f . I get this : laps1:~/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc # make load make -C /usr/src/linux-2.6 SUBDIRS=/root/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc

After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman

I upgraded from 1.0.9 to 1.2.1 My IVR which worked perfectly on 1.0.9, now hangup with no reason (at least I could not find a cause) When this hangup happen, I can read: == Auto fallthrough, channel 'IAX/user-20' status is 'BUSY' This happening also with ZAP channels I'm really disappointed with 1.2.1, what is benefit from upgrade if I must spend couple days to get my system

Hi!I've installed Asterisk@home and I have a ISDN card,(Cologne Chip Design GmbH ISDN network controller [HFC-PCI](rev 0.2) This is how I installed bristuff: how to install hfc card after unload asterisk and amportal whit amportal stop type "setup" unselect zaptel in system service... and set the lan --->reboot<--- cd /usr/src wget

Hi Last friday I have upgraded to Asterisk 1.2.7.1 (bristuff-0.3.0-PRE-1p.tar.gz). Since that I have a problem with my Asterisk box. I am receiving these messages: May 24 09:30:11 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). May 24 09:30:12 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2).

Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a x86_64 box (I guess nobody is using x86_64 platform or is able to fix this themselves?) First of all when bristuff is downloaded and compile is started it appears that the bristuff Makefiles are badly broken. The asterisk Makefiles all do see to find the kernel sources on a RHEL4 box in the proper directory, the pure

I looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or uncomment this: [options] ;silence_suppression=yes And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan

I have a problem with Asterisk-bristuffed using a zaphfc card. I am located in the Netherlands, so I have an ISDN line from KPN. When I start Asterisk, and plug in the ISDN line, everything works perfectly for about 5 minutes. And then the ISDN line is down for 1 minute, and after that minute, the line comes back up and works for another 5 minutes. Every time the line goes down I get the error

Hi, I have debian etch 4.0 machine (2.6.18) with two TW-ISDN PCI (Hfc) cards. I use bristuff-0.3.0-PRE-1y-e (asterisk-1.2.17,libpri-1.2.4,zaptel-1.2.16). I also have patched zaphfc with zaphfc_0.4.0-test1_florz-13.diff.gz (I load module: insmod /usr/src/bristuff-0.3.0-PRE-1y-e/zaphfc/zaphfc.ko modes=1 debug=1). So i want to test two cards and make loop between them. So one card would be NT,

Hi, I'm setting up an asterisk server with this hardware configuration: AMD Athlon 1000 Mhz 256 MB ram 3ware ATA raid controller 2 * Ethernet controller 2 * ISDN HFC controller One ethernet controller is connected directly to the internet (public IP) One ethernet controller is connected to the internal lan One ISDN controller is connected to the public telephone network One ISDN controller

Regarding my previous post: "Playback audio before answered by a queue member" I added a ResetCDR() command at the middle: exten => XXXXXXXXXX,1,Background(audiofile) ;answers the channel immediately exten => XXXXXXXXXX,2,ResetCDR() ;clean slate exten => XXXXXXXXXX,3,Queue(Qname|tdn|||) ;new answer time written Looking at the CDR, the billsec is no longer the same as

Im using bristuff-0.2.0-RC8g with two HFC-PCI controllers. Inbound calls work just fine, but when im dialing out asterisk shows: -- Executing Macro("SIP/8010-20a1", "dial_out|xxxxxxxxxxx") in new stack -- Executing Answer("SIP/8010-20a1", "") in new stack -- Executing SetCallerID("SIP/8010-20a1", "xxxxxxxxxx") in new stack

I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't

Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between