similar to: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit.

Hello. Scenario: 9 servers connectec to each other over IAX trunks. Users used to call to remote extensions and remote conferences (meetme) via IAX. Problem: all extensions from one server (just one) when try to attend remote conferences had problems with PIN validation. If they use their local conferences the problem doesn't occur. All servers use same OS (Ubuntu 10.04 64 bits), same

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

Are there any known problems converting dtmf from oob over iax2 to inband over rtp/ulaw? Obviously it works when converting to inband over pri/ulaw et al, but how about rtp? I've got packet traces that confirm that 2833 packets are properly generated when I have 2833 configured for the rtp link, but the other side seems to be ignoring those packets. So I tried inband on that link; nothing

Hello, I am searching for a DTMF issue on my setup ( 2 years and counting ), and I am wondering why rtp.c has code to mute DTMF ( the rtp->dtmfmute variable ), but this same mechanism does not exist in dahdi. I am sending a DTMF over SIP w/ RTP & RFC2833 to the asterisk box with the dahdi card. The dahdi card sends it out on the PRI line. Trouble is, the DTMF is echoed back and the

On Wed, Feb 4, 2009 at 4:00 PM, Gregory Boehnlein <damin at nacs.net> wrote: > Hello, > Is anyone running Asterisk 1.4 w/ RFC2833 to Level3's SONUS network? > We are unable to get reliable RFC 2833 DTMF working, and have instead had to > use G711ULAW w/ INBAND DTMF to get around the issue. Looks like an issue on > the SONUS side. > > Anyone else have this

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 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 again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took

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 a TDM22B (TDM400 PCI + 2xFXO + 2xFXS). One FXO is connected to the PSTN. When Asterisk places a call, it dials using DTMF. If I listen in on the line during the dialing, there is a roughly one second pause between the penultimate and the final digit -- proportionally much longer than the pause between the preceeding digits. This accounts for about 25% of the time taken to dial. Anyone

Good Morning, I've recently gotten Asterisk installed and configured our IVR using FreePBX. Things seem to be going well except a few of our inbound callers are ending up in the wrong place when trying to connect to a specific extension. The example I had this morning was someone trying to call extension 212 and getting connected to the Sales queue which is option 2 on the IVR. I looked in

I get the following on the console whenever I request an outside line (9+) as soon as the 3rd DTMF digit is pressed on the (zaptel interface) telephone set. NOTICE[1226062640]: chan_zap.c:3587 zt_read: Fax detected, but no fax extension. I've searched the mailing list for "FAX detected" and there is nothing mentioned about getting this message during outgoing calls. The error

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

Dear friends, As I commented some while ago in the list, occasionally when DTMF Tones are sent, they appear in RTP Payload and in Events too, producing duplicate tones being recognized. This behavior happens in Asterisk as well as in Gateways such as Cisco, for which we had the opportunity to observe the error and extensively debug it. We ended up recognizing good digits by adjusting audio gain

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough

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