similar to: Damn DTMF Beeps on my calls

I have a very annoying problem that we hear on our end, but the other party doesn't hear. There are random beeps and echo type noises that occur. They are present during voicemails, and present on my end during calls. Is anyone experiencing the same deal? I have asked this a number of ways on the list, and never get a response... Thank you. Sean Garland Mount Shasta, CA

Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility,

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

Hi all, I use a GSM device to send dtmf on my asterisk system (via SIP). the codec I use is ulaw (or a-law). dtmf mode is INBAND. relaxmode is on. but most of the case, I 'missed' some DTMF or I 'double' one. as anybody as seen this before? is there any way to prevent this thanks

I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

Hi Tom, Thank you for your informative and helpful reply. I had considered using the relaxdtmf setting but held off this due to not using any physical connection hardware -Asterik uses both SIP in and out from an upstream provider (Gradwell.com). Is it still possible to set this when using SIP trunks only and not physical hardware? The box does have a Digium ISDN card but the ISDN is no longer

Hi, I am a newbie to Asterisk; need help understanding three-way conferencing & call-transfer features implemented over standard extensions i.e. on a TDM800P card (4 FXO + 4FXS) In Asterisk I have observed that if an extension is already participating in an active call (e.g. Ext A & Ext B communicating): 1. An incoming call to one of these active extensions would be presented with

Ah I see, in theory it's possible then. We don't have any IVRs or anything which requires key presses, there isn't even voicemail on this particular phone system so I don't think it will be too much of a problem. I've also updated the firmware on the Cisco phones that have had the issue, just to see if that solves the issue but as it's been going on for a while, I'm

I still have problems with beeps and clicks on all my calls. I have polycom sip phones. I also can hear the beeps and clicks on some of my messages, which would lead me to believe that it is more of a decoding problem on the zaptel card. Any ideas? Thanks Sean Garland

freebsd quite current ports tree 1.01 asterisk spa3k at 2.0.11(GWg) for calls in from the pstn side of an spa3k to asterisk, i get sporadic short beeps. they are not related to sip re-reg time, which is all that has occurred to me so far. calls in from the fxs side of the spa3k and out through nufone do not exhibit the beeps. calls from the fxs side of the spa3k out the fxo side do have the

Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It

So, just today I noticed a problem with kernel 2.6.18-194 (CentOS 5.5) on several Intel DP965LT systems; the system beeps (such as terminal beeps) are no longer passed through to the external speakers. This is a problem because in our situation the boxes are distant from their monitor/keyboard, the system speaker on this motherboard is extremely weak, and there are no system speaker header pins

Hi, I'm trying to implement a Voice Drop service within Asterisk dial-plan. The service is supposed to work as following: 1. A initiates a call to B 2. The call is answered by B's answering machine 3. A hears the answering machine's greeting and the recording beep 4. A speaks a few words into the recording to personalize the message 5. A presses some DTMF keys (say, '##') to

Hi, I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO. I'm using a VTech cordless that makes three short beeps when someone another extension is picked up, presumably this lets you know if someone is trying to listen in.. Everything works, except the VTech now makes the three beeps everytime you try to use it, even if another extension is in use. It seems as though the

I found this old post using a Google search for DTMF tones heard during an Asterisk call. The calling party does not hear them, only the called party. We are experiencing this sporadic DTMF "beep" followed by a momentary silence on a random number of our calls. This happens on trunk to SIP and SIP to SIP calls, so it's not PRI related. Since this thread is dated 2003, and the

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

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the *

OK here is a possible tricky one. I have a rocom door entry system which connects to an FXS port on my TDM400P card. When the door button is pressed it initiates an 's' extension which dials a number of SIP extensions. When a SIP phone is picked up the user can speak to the person at the door and press the 7 digit which sends at DTMF tone to the rocom unit opening the door. All this