search for: toneduration

...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 toneduration I can control duration of tone per digit. But I can't find a way to do both at the same time Application sendDTMF simply ignores the value set in toneduration and sends DTMF at some default value, which I don't know what it is, but it is obviously not 50ms because the hardware can't re...

...nes[] = { ZT_TONE_DIALTONE, ZT_TONE_BUSY, ZT_TONE_RINGTONE, ZT_TONE_CONGESTION, ZT_TONE_DIALRECALL, }; int main(int argc, char *argv[]) { ZT_DIAL_OPERATION dop; struct wctdm_regop regop; struct zt_dialparams dps; int fd,ctlfd,toneduration; int res; int x; if (argc < 3) { fprintf(stderr, "Usage: fxstest <zap device> <cmd>\n" " where cmd is one of:\n" " stats - reports voltages\n"...

...plication, I can control the duration between two DTMF digits, but I can't find a way to control the duration of the digits themself. Did search on the Internet and found out that I can change it in the asterisk source files and recompile asterisk. Wiki also says that it can be controlled using toneduration option in zapata.conf, but it doesn't seem to work. Does anybody know if there is an easier solution to achieve this without re-compiling Asterisk. I have to test it with various different durations and don't want to recompile asterisk every time I change the duration. Thanks, -- Zeesha...

...ms with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do anything that we can measure. We have also tried setting channel.c parameter #define AST_DEFAULT_EMULATE_DTMF_DURATION 200 to several different values but none seem to alter DTMF duration at all. Does anybody have a clue where we can hardcode DTMF duration for tones...

...io in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes answeronpolarityswitch=yes usecallerid=yes cidsignalling=bell cidstart=ring ;hidecallerid=yes ;hidecalleridname=yes ;waitfordialtone=yes ;mwimonitor=no ;mwilevel=512 ;mwimonitornotify=/usr/local/bin/dahdinotify.sh ;mwisendtype=rpas,lrev callwaiting=yes ;restrictc...

I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN

Hi, I'm new to this list I'm developing asterisk application where users can call and control volume up and down in music player. Problem I'm getting is if users press 222228 in fast speed, system will process all those 2s and then process 8, so there is few seconds ( around 4-5) processing key press 8 , therefore users will feel unresponsiveness in system.(in other words users will

...0 pickupgroup = 0 immediate = no echotraining = yes echocancel = yes echocancelwhenbridged = no facilityenable = yes musiconhold = default ;overlapdial = yes overlapdial = no immediate = no txgain = -4.0 rxgain = -4.0 signalling = pri_cpe channel => 1-15 ;channel => 17-32 channel => 17-24 ;toneduration=100 toneduration=300 ;relaxdtmf=yes Thanks, -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

...beginning of the sound message (without waiting to complete it), then asterisk might read the digits duplicated (specially first entered digit), how can I resolve it? Does playback resolve my problem? Any solution for Background to avoid this behaviour? Note: I set the relaxdtmf=yes and I made the toneduration=500. Looking to hear any advise :) - Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We?ve already added "w" before the dialed number with no results, is there any way to solve, is it a bug thanks

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

...I can't do much with it, other than find the right settings. I have on the first one relaxdtmf=yes - relates to old issues too as far as I can tell rfc2833compensate=yes - this only appears to matter for inbound I'm not sure these do anything useful From what I can tell it could be the toneduration, but don't know what it should be, and while technically its probably the IVR being fussy that doesn't help me and I want to see why one system works and one doesn't This is dtmf debug from an iax handset sending digit 4 [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set ch...

...hone for several times the card just detects one or two digits randomly.so now i can't use any voice menu on my box with this card. i have tried the following scenarios: - the card with / without vpm module has the same dtmf detection problem. - relaxdtmf=yes/no didn't solve the problem - toneduration=300 / 350 / 400 didn't help also. - vpmdtmfsupport=1 / 0 didn't solve again. what else could be the possible cause for this problem? please help! - paradise dove

Hi I have a problem with dtmf recognition an analog lines connected to Sangoma A200. The digits (in most cases the first one) are doubled and so my IVR is useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but nothing worked. I also noticed one thing it only happens during the background application: exten => s,1,Background(soundfile) exten => 111,1,Dial(SIP/111) exten => 122,1,Dial(SIP/122) it never happens in WaitExten application: exten => s,1,WaitExten(25) ext...

...he alarm system uses Contact ID protocol. My problem is that the negotiation fails and I think that the problem is that "kissoff tone" is cut and the transmitter doesn't recognize it. Maybe the asterisk tone duration isn't long enough. I'm thinking about increasing the "toneduration" value in zapata.conf. or changind DTMF tone frecuency. Does anyone deal with a similar problem? What are the optimal values? Thanks Regards

...ay , I have configured TDM410P (asterisk 1.6.x) on Cent OS 5, but dahdi take some time to hangup the call when playing the IVR..(it will send the hangup signal after finishing the IVR promt..) is there any specific setting to avoid such incidents ? iam using busycount as 3, signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes echocancel=128,param1=32,param2=0,param3=14 echocancelwhenbridged=yes echotraining=yes echotraining=800 busycount=3 hanguponpolarityswitch=yes ringtimeout=8000 group=1 context=incoming immediate=yes jitterbuffers=4 jbenable = yes echocancel=yes...

...:07] VERBOSE[1935] logger.c: << [ TYPE: DTMF End (1) SUBCLASS: 8 (56) ] [DAHDI/6-1] [Jun 17 12:39:08] VERBOSE[1935] logger.c: << [ TYPE: DTMF End (1) SUBCLASS: 0 (48) ] [DAHDI/6-1] Our telco provider has told us that signalling is sent via inband mode, so I've messing with "toneduration", "relaxdtmf" parameters on chan_dahdi.conf, but I don't get better results... As I have read in Asterisk lists the echo canceller module can be configured to detect DTMF tones via hardware, but it's disabled by default. Maybe activating it I'll get an improvement on...

...application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: SetMusicOnHold [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Goto [Sep 21 18:11:48] DEBUG [8536] chan_dahdi.c: Took DAHDI/10-1 off hook I use the headset Zox TS19. I tried changing the value of toneduration = 100 but did not work. Anyone know how I can solve this problem? thank you very much. Marcus Vin?cius. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100921/89e37f90/attachment.htm

...:07] VERBOSE[1935] logger.c: << [ TYPE: DTMF End (1) SUBCLASS: 8 (56) ] [DAHDI/6-1] [Jun 17 12:39:08] VERBOSE[1935] logger.c: << [ TYPE: DTMF End (1) SUBCLASS: 0 (48) ] [DAHDI/6-1] Our telco provider has told us that signalling is sent via inband mode, so I've messing with "toneduration", "relaxdtmf" parameters on chan_dahdi.conf, but I don't get better results... As I have read in Asterisk lists the echo canceller module can be configured to detect DTMF tones via hardware, but it's disabled by default. Maybe activating it I'll get an improvement on...

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has