similar to: no dial tone

Hello all, I'm trying to setup a AVM C2 card. I have read the kernel requirements for this card. <M> CAPI2.0 support [*] Verbose reason code reporting (Kernel size +=7K) [*] CAPI2.0 Middleware support (EXPERIMENTAL) <M> CAPI2.0 /dev/capi support [*] CAPI2.0 filesystem support <M> CAPI2.0 capidrv interface support My problem is when I make a "make

Hello, I?m trying to compile the Fritz CAPI module for Debian stable, following the steps related in http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install But I always get the same error, debian-asterisk:/home/ismaelg/fritz# make (cd src.drv; make CARD=fcpci) make[1]: Entering directory `/home/ismaelg/fritz/src.drv' cc -c -DMODULE

Hi all, I have a Diva Server V-BRI-2 card, which support, as written in reference guide: Extended tone processing (human talker detection, generation and detection of country-specific tones) I would like to detect human speech and fax tone with asterisk. I think that the diva card transmit a DTMF code when detecting voice, but chan_capi doesn't receive this DTMF code. I verbose it

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

Hi, I have a strange problem with our Asterisk installation. Outgoing calls are handled by the following lines: exten => _0[2-9]X.,1,Set(CALLERID(num)=09999403${CALLERID(num)}) exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} = 0999940321]?099994030:${CALLERID(num)})}) exten => _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr) exten =>

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

Oh geez nobody respond, I guess I have to be more descriptive. I'm install Asterisk and FXO FXS Lite dev kit from Digium everything has installed and I am getting an error message that my sound card is not enabled. I'm using a Dell Optix (whatever). Everything seems to be in order. I connect an analogue telephone to the FXO port? and try to dial. Dial tones work however not dial tone.

Hi Folks. I see that can put 0 to call out using a x101p (zaptel) or even a pstn service. Thats great, but when press the 0 i just dial then the numbers to call out. There is any way after hit 0 (ear) the line sound ?? I know it's just a style way put some users, really like it !! So after hit 0 to call for example a pstn the user will ear the line sound to dial out. I read lot's of

It works!!! Thank you so much for your help! Sent from my iPhone > On Feb 3, 2020, at 3:47 AM, Rui Barradas <ruipbarradas at sapo.pt> wrote: > > ?Hello, > > You can solve the problem in two different ways. > > 1. Redefine storage1 as a matrix and extract the aic *in* the loop. > > storage1 <- matrix(0, 4, 4) > for(p in 0:3){ > for(q in 0:3){ >

Hi, I've looked in various FAQs and web pages for this info, but just can't seem to find it. When two tones of two different frequencies sound equally loud, what's the (rough) relationship between their power? This is for percussive sounds in music, but I assume it's roughly the same for all sounds. The ear seems more sensitive at high frequencies. For example, when you

How can I bring the Caller ID when the calls enter call queue and answer by X- lite or kphone? I've tried many configuration but no luck that it only shows the AgentLogin's exten.. Thanks! R Wong The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission,

I've been trying to play the default music on hold file, but no luck yet. here is my configuration: extensions.conf [incoming] exten => s,1,Dial,Zap/2|10 exten => s,2,Voicemail,u34 exten => s,102,Voicemail,b34 exten => 34,1,SetMusicOnHold,default Musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random =>

I am nowaware that I should not post this type of questions on this group. However, Iwould like to have some clarifications related to the response you've?alreadyprovided. The code you provided yields accurate results, however I still haveissues grasping the loop process in case 1 & 2. In case1,?the use of?"p+1" and "q+1" is still blurry tome? Likewise "0L"

I am routing outgoing calls through a sip gateway. The calls go through no problem, however the ringing in the callers ear begins as soon as the last digit is dialed. This has two nasty side effects. First, the caller hears 1-2 more rings than the callee. Second, and more importantly, if the callee's line is busy, the caller continues to get hear ringing, even though the gateway has

Hi ml, I have a problem related to call parking. When on my X-Lite try to parking a call dialing #700 I don't obtain anything. I can only ear dtmf tones during conversation but not other happens. I also read in some post that only pressing # should place call in hold state but this doesn't happen on my system. Can someone help me? Thanks. Marco

On Wed, Oct 04, 2017 at 09:44:44AM +0000, ismael mondiu wrote: > Hello, > > I'd like to test if 3.10.6 version fixes the problem . I'm wondering which is the correct way to upgrade from 3.10.5 to 3.10.6. > > It's hard to find upgrade guides for a minor release. Can you help me please ? Packages for GlusterFS 3.10.6 are available in the testing repository of the

Hello I am trying to extract AICs from an ARIMA estimation with different combinations of p & q ( p =0,1,2,3 and q=0,1.2,3). I have tried using the following code unsucessfully. Can anyone help? code: storage1 <- numeric(16) for (p in 0:3){ ? ? for (q in 0:3){ ? ? ? storage1[p]? <- arima(x,order=c(p,0,q), method="ML")} } storage1$aic [[alternative HTML version deleted]]

Hello all, Recently I bought a TDM02B digium card to conect to the PSTN. We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try to start asterisk, the box hangs. Someone have the same card running with asterisk in a similar machine? Could you tell me your box hardware details? Thanks for your time, Ismael Gil.

Hello all, I am planning to connect my Asterisk with the FWD and/or iaxtel networks. Two mounths ago, I just used the iaxtel network, and i remember I have trouble with this network, I can not place a call. The service do not wotk. With FWD I alwais can place a call, I never get an error from this network. But my experience in both are very short, just a few test time. Could somebody

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither