similar to: newb question regarding DTMF

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't

I have this silly problem : When I place a call, being either to an extention or to an outside line, DTMF signals are ignored by Asterisk. This is serious because I can't even transfer calls (#) or park them (#70). When I receive a call there's no such problem. When I recover a call from parking (71) all goes OK too, and so goes call capturing with *8... I already tested dtmfmode=inband,

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

We are developing an IVR application and when I am testing locally on my machine using a softphone (iaxcomm) the digits I press for GET DATA work every time. I am testing with a local extension that goes right into my routine. However when I try to call in to the system using an analog or cell phone GET DATA drops some digits that are pressed. There doesn't seem to be a pattern to which

Hi I believe this is due to the way US phone systems work, however I'm going to ask anyway. In the UK there are several providers who provide national rate PSTN => Voip gateways which are free to receive calls on, (for the recipient), the caller pays the cost of calling. E.g 0844 0870 etc. I am looking for a US provier who offers the same sort of system. I don't call the US but I

Hi, My system is like this currently: ATA-186 <-> *1 <-> IAX2 to Europe <-> *2 <-> i4l <-> voicemail at cell provider When I dial up to my voicemail at my European cell phone provider I can't press '#' to get into their menu. It seems like it just ignores any DTMF tones or doesn't get them. When I call a human on the other side of the i4l they

Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which

Hey - I know there's some other people out there that have the 9133i ... has anyone gotten the DTMF tones to work after the far side picks up? I didn't have any problems out of the box with my SPA-841 phones... the aastra has been nicer so far, but I can't seem to get it to dial the touch tones after an auto-answer device picks up on the far side... I googled, to no avail. -Karl

Dear All ~ I have * setup & running ok (with two Wildcard X100P's to PSTN). I also have two analog phones connected into same through a SIPURA 2000. These work fine, except that when I call out through PSTN & try to send DTMF tones to (say) a remote PBX to dial an extension, the gain seems to go wild (high), and the DTMF tones are not recognized at the other end. I tried setting the

Hello, I am fairly new to asterisk. I am currently using asterisk as a more convenient sip side voicemail system. My problem: I have cisco 7960 phones whose out of band dtmf tones are recognized properly(when dtmfmode=rfc2833) by asterisk but whose in-band dtmf tones are recognised poorly(when dtmfmode=inband) . For example 7999 comes out as 799999, 4242 comes out as 442422 ... etc I

Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes. I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I configure dtmfmode=rfc2833 (I've tryied inband and info). Asterisk seems not to "see" the tones. Could somebody help me? Thanks

Hello We have discovered a problem with DTMF on Asterisk. We have a setup with a T1 from PSTN going into an Asterisk box, and then out again on T1 and into a normal PBX (EADS) We use it to record all calls going to/from the PBX. The problem is that when we record the calls (with MONITOR command), DTMF tones gets obscured, and is not understood in the other end, if we dont Monitor, there are no

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

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

--On Monday, January 19, 2004 11:01 AM -0500 Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: <SNIP'd from the "ADSI phone vs. IP phone" thread> > I'm looking at ADSI phones simply because I don't have to re-tool my > entire building; I can use the existing phone network and (I think) get > all the functionality I need with the (far) cheaper

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

While most cell phones are fine, some cell phones don't seem to produce DTMF digits that Asterisk/Zapata will detect. One of our salespeople has an AT&T model that never gets any digits through. Is there a known solution? I understand, of course, that with cell phones the DTMF tones are actually created at the base, and that pressing the key longer usually does not result in longer

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer

Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I