similar to: DTMF being cancelled

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

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

Hello, On the Polycom IP 500 Phones, when I press the mic mute button, the mic on the speaker or headset goes muted. However when I press the mic mute button again, the call is terminated by asterisk. Asterisk shows a: WARNING[1236268096]: File channel.c, Line 1296 (do_senddigit): Unable to handle DTMF tone 'f' for 'SIP/####-####' I am using reinvite=no on the phones. After

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 all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards: a TE420 w/Octasic and pri_net

Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is

Does anyone have any idea what would cause distorted audio but ONLY for DTMF tones coming in over our analog lines. (The analog interfaces are X100P's). I have carefully adjusted the gains in the zapata.conf using a local test line after trying various settings with no gain or just random gain settings. RelaxDTMF has no effect. I set up a monitor command in my dial plan to capture

--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

I just wanted to see if any one else has seen this or could help point me in the right direction on this problem. I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I

> I have seen the following effect in Asterisk, though: where > it converts > an inband DTMF (eg coming off a Zap channel) into an > indication, it mutes > the audio where that tone is. But sometimes it leaves a > teeny bit of the > tone behind. > > If you take such a call over say IAX to somewhere and then > back out a Zap > channel, you end up with the

I am using Asterisk 1.06 Stable. When I dial my Mobile Number to check Voice Mail or my Bank Account Phone Access Number, the IVR System on the other end asks me to enter *2378 to transfer to an attendant. But When I press *2378, Asterisk tells me that it cannot transfer the calls and gives an error on CLI saying Extension '' does not exist in the dial plan. What is the trick to make

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

modern technology, shorter durations could work. Most phones of all types don't make a standardized tone burst but produce tones only while the button is pressed. Fast "punching" will produce short tones. On the other hand, a redialed number will be very well formatted. Reliability of TT data transfer for audio applications (over the phone voice mail, credit card, IVR, etc) would

We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that

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

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

Hi, we have an Asterisk server basically passing on calls using the Dial application. In the pjsip endpoint settings, the dtmf_mode is set to audio. This works with most calls. However, there is a scenario where DTMF tones don't get forwarded the way I would expect them to get forwarded. A: Caller without RfC4733 support B: our Asterisk, version 17.6.0 C: Another Asterisk, with RfC4733

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

Hello, I am using Asterisk SVN, a cellular phone, and chan_mobile to run a small home PBX with two analog telephones connected to a Linksys ATA using SIP. It works great (except for some Bluetooth adapter bugs that I am still trying to beat...seems the misaligned audio detection still needs work), but I have encountered an interesting issue. If I am using an automated system that accepts input

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP providers are not very good at doing that suppression (leaving audible clicks, or failing to suppress the tones