similar to: DTMF transmitted over IAX2 coming out as clicks at the other end

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems

for the particular configuration of software/hardware that connects to my asterisk pstn gateway I need to do something like the following : [...] exten => _X,3,Dial(CAPI/02xxx.b${EXTEN},60,M(senddtmf)) [...] [macro-senddtmf] exten => s,1,SendDTMF(*) but the DTMF must be sended to the caller channel, and not the called : SIP -> * -> ISDN SIP calls some ISDN number, when ISDN picks

Hi folks, My stack of trusty FreeBSD servers always seems to be growing, and it's getting to the point where the daily and security output mail is too much to make good use of. I'm looking for suggestions for log monitoring and aggregation tools, especially from a monitoring-for-security perspective. If I had to imagine an ideal system, it would be a central server that securely

Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recognized. Using command sendDTMF, I can control inter-digit duration, and using

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have

I am trying to send a DTMF digit automatically every 15 seconds to keep a call connected to an alarm panel. I tried using the dial command L and recording a dtmf tone for the beep, but obviously that didn't work. Does anyone have a suggestion for merging the L option and the sendDTMF or the D option? Any other suggestions would be appreciated! Thanks! Paul Gentilini

Hi list, I would like to know of a possible way to dial a pstn number with an extension . Let the number is 56626965-234 so now i wanna dial 56636965 then wait for some time and dial the extension 234 to reach a particular person.I am afraid that i could not figure it out. I am trying in this way.. [outgoing] exten=>_566X.,1,wait,2 exten=>_566X.,2,Dial(${EXTEN})

I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551212 at sip.com,60,rD(12345#) The dtmf is sent too soon. I tried inserting 'ww' but that was just sent. I tried G: exten =>

I'm trying to record a Meetme conference to disk, but the Monitor application doesn't seem to play nicely with Meetme. In extensions.conf, I have this: exten => 1000,1,Answer exten => 1000,2,Monitor exten => 1000,3,Meetme This starts up the monitoring OK, and it records the prompts that Meetme gives, but as soon as the user enters the conference, the -out WAV file stops

Hello, I did this post a long time ago but never solved the problem, so i'm trying again after something like 10 months, hopefully i'll find someone that found a solution ;-) When i call an external number that sends audio before call has been answered (like some PBX of public offices do here in italy), strange things happen: I'm using chan_capi, with Early B3 active, i can listen

I'm having an issue sending DTMF to cisco dialing this extension I should hear the dtmf tone RTP playload 101 has been sent to the cisco phone, but no audio. in the dialplan exten => 8603,1,Answer(1) exten => 8603,n,sipdtmfmode(rfc2833) exten => 8603,n,SendDTMF(1|100) exten => 8603,n,hangup() sip.conf dtmfmode=rfc2833 SIPDefault.conf I did play with all possible settings for

Greetings All! The Asterisk Call Manager works great. But I have one question for anyone who has used it. I cannot get the system to send some DTMF tones down the channel once the call has been made. Below is the script I am using to make the call, and start recording the channel. I am starting to make a system the will use asterisk to become an automatic random quality monitoring system

Hi I'm trying to implement a Callback mechanism whereby I generate a Call file and connect an arbitrary extension with my cellphone (via a SIP Channel). If I create a .Call file that connects the channel "SIP/12345678@Provider.net" with a local extension/context I get some weird issues with DTMF tones. I've set dtmf=2833 and the codec in use is G711a. For example - I create

Hello, I setup a door open system with a basic DTMF card. The card is connected to an Sipura/Linksys 3102 FXS port and is powered by this port. My problem is that when I send a call with Dial() command, channel has to be answered before receiving DTMFs, what my card does not. Is there a way to autoanswer those type of calls or to send DTMF on an non answered channel or another solution/idea

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over SIP trunk from which calls get routed to third server (C) (1.6.0.9) via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP

I am having trouble with a DTMF-based application on Asterisk 1.0.3. Specifically, when two IAX2-based devices are talking, when they send DTMF to eachother, the other side only hears clicks, and maybe a millisecond of DTMF tone, but not any real duration. Furthermore, when one IAXy device calls the Echo test program, we can hear our echo, but when we punch DTMF in, we get the same effect

This 2 line code is doing what I wanted. exten => 200,1,voicemail(200) exten => 200,2,Hangup What I've been told is that they want the 20 year old phone system to light up the message bulb. (yea, a filament bulb, not an LED) To do this you pick up on the line that goes into Asterisk and do a: exten => 200,1,SendDTMF(200w#86) But I don't know the path to take to get that

Hi all, I've seen a couple of posts recently from people who are doing something with Asterisk and a T100P and a Nortel PBX. However it's not clear exactly what they are doing. Does anyone know if it's possible to use a T100P or T400P as a DSX-1 interface to connect to a T1/PRI CO module in a Nortel PBX? We have a Nortel Norstar Modular ICS PBX and I'd like to plug its CO