search for: relaxdtmf

Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffre...

.... For getting DTMF I have use one opensourse application which gets events from asterisk server and store into database. And after that I made my own script to gets these DTMF keys and play some IVR into the conference. For conference I am useing Konference module. I have check sip.conf and set *relaxdtmf=no* I am using Dahdi and set chan_dahhi.conf as below [channels] resetinterval=never context=incoming usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancel...

...he CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them what they changed (fat load of luck getting that question answered anyway). Everything was working fine with dtmfmode=inband and relaxdtmf at the default, now I've tried all 6 valid combos of those two settings with no change. This is on asterisk 1.2.27 that's been working fine in production for about 3 months now. Here's the section from sip.conf (the way it had been working all along): [viatalk] type=peer secret=(yep...

...nes calling via the PSTN (we use a SIP trunking carrier and do not handle the PSTN interface ourselves); we've seen it from land line phones via the PSTN, and have even seen it internally from our own Snom SIP phones. dtmfmode=auto but we have also tried setting it to rfc2833 and we have tried relaxdtmf set to both yes and no. We are running Asterisk 1.6.1.6 on CentOS 5.3. We really don't know what more to do to fix it. Googling shows that others have had this problem but I haven't seen a clear resolution other than playing with relaxdtmf. How do we solve this problem? Thanks - John --...

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 lon...

...adjusting this setting will (for you) do nothing, since you don't use the dahdi channels for transport. See this discussion, which I found after I posted my first response: http://www.voip-info.org/wiki/view/Asterisk+DTMF Particularly this sentence: "Note: Asterisk 1.4 now also has the relaxdtmf= setting available in sip.conf." The big question for you is going to be, does your system need to recognize inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems doing that? Thomas M. Peters | Systems Administrator | tpeters at mcts.org Desk: 414.343.1720 | Helpdesk: x3...

...n=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 loadzone=us defaultzone=us ======================== zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] busydetect=1 busycount=7 relaxdtmf=yes callerid=asreceived context=default signalling=pri_cpe usecallerid=yes transfer=yes echocancel=yes echocancelwhenbridged=yes switchtype=dms100 rxgain=0.0 txgain=0.0 immediate=no signalling=pri_cpe group=1 context=default channel => 1-23 busydetect=1 busycount=7 relaxdtmf=yes ca...

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten => _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallow=all allow=g729 ;allow=ula...

...valid option, bcos there is no extension with 223. I had to do this bcos I got feedback from many users saying that when reaching their extension they get these invalid options, so using my phone was the only way to replicate it. Further contacting Digium support they asked me to enable. the relaxdtmf=yes option in zapata.conf. I did & still the same issue. What is this a bug to live with or issue which has a solution. ================== Asterisk DEBUG Message ================== -- Playing 'custom/Greet1' (language 'en') -- Invalid extension '223&...

...,Congestion /etc/capi.conf contains : c2 c2.bin DSS1 - - - - P2P c2 c2.bin DSS1 - - - - P2P /etc/asterisk/capi.conf contains : [general] nationalprefix=0 internationalprefix=00 rxgain=0.9 txgain=0.3 ; interface sections ... [ISDN1] isdnmode=DID incomingmsn=* controller=1 group=1 softdtmf=on relaxdtmf=on accountcode= context=capi-in holdtype=hold echocancel=yes echotail=64 bridge=yes callgroup=1 devices=2 [ISDN2] isdnmode=DID incomingmsn=* controller=2 group=1 softdtmf=on relaxdtmf=on accountcode= context=capi-in holdtype=hold echocancel=yes echotail=64 bridge=yes callgroup=1 devices=2

...p; 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 SIP2000 to use inband dtmfmode (as opposed to auto), and likewise in sip.conf, but no success. btw, I've also set relaxdtmf=yes in zapata.conf since inbound calls sometimes seem to have trouble dialling extensions. A soft IAX phone (e.g. DIAXPhone) works ok, so I suspect my SIP2000/sip.conf setup, but can't see what I'm doing wrong. Christina.

...PCI) Card 0 Span 1" ; This T1 is attached to in-house CUST 3 System ; language=en rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =>em_w group = 2 channel => 1-24 ; Span 2: TE4/0/2 "TE410P (PCI) Card 0 Span 2" ; This T1 is attached to inhouse CUST 10 System ; language=en rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocan...

...callerid=yes hidecallerid=no callwaiting=no usecallingpres=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes immediate=no musiconhold=default ;;; INCOMING COPPER X100P signalling = fxs_ks ; x100p group=1 echocancel=yes relaxdtmf=yes callerid=asrecieved context=copperincoming channel => 1 ;;; INCOMING VONAGE X100P signalling = fxs_ks ; x100p group=2 echocancel=yes relaxdtmf=yes callerid=asrecieved context=vonageincoming channel => 4 ;;; INTERNAL TDM20B signalling = fxo_ks ; tdm20b echocancel=yes r...

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 seem to solve the problem. If I turn relaxDTMF off then most incoming calls from Cell Phones cannot navigate our menu and I get doubled digits showing up in the log. With it enabled 99% of calls w...

...ber, its taking the first digit of USA number twice. For eg: If I enter 17187773456, its taking as: 117187773456 If I enter 917187773456, its taking as: 9917187773456 Its taking my input USA number correctly for sometimes and call is connected to my mobile. I tried by changing the value of "relaxdtmf" from yes to no and vice versa. Here I am sending my config files. Please tell me the solution. Extenstions_custom.conf contents: [custom-CLID] exten => s,1,Answer exten => s,2,DigitTimeout(5) exten => s,3,ResponseTimeout(10) exten => s,4,Authenticate(1234) exten => s,5,DISA(...

...oks like this: [incoming] s,1,Dial(Zap/2,15) s,2,Voicemail(u10) s,3,Hangup My voicemail.conf looks like this: [default] 10 => ,Shawn Wilson,sbwilson@sympatico.ca and zapata.conf looks like this: [channels] context=incoming signalling=fxs_ks language=en echocancel=yes echocancelwhenbridge=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 channel => 1 context=local-access signalling=fxo_ks language=en echocancel=yes echocancelwhenbridge=yes musiconhold=default callerid="Sample Zap Phone" <10> relaxdtmf=yes txgain=1.5 rxgain=1.5 channel => 2 If anyone can figure this out it would be g...

I am having issues with a TDM2400P. It appears when the ZAP channel dials out, it randomly chops the first digit off of the number. I have tried relaxdtmf=yes, turning up and down the txgain, turned off and on the echo cancellation, generated new zaptel (with updated spinlock.h)... I am at a loss. Can someone please offer some help? Thanks. TJ

> I don't see anything obviously wrong with your configs. > You don't want relaxdtmf. That can cause the problem, not fix it. Hi Eric, at the begining -> Thanks for your help. relaxdtmf is not written in my config, so it should be at the default, i guess i remember default is yes ? However, the dtmfmode should be the same, i think so, too, but my SNOMs working pretty well...

> I don't see anything obviously wrong with your configs. > You don't want relaxdtmf. That can cause the problem, not fix it. POST 2 --> got no response Hi Eric, at the begining -> Thanks for your help. relaxdtmf is not written in my config, so it should be at the default, i guess i remember default is yes ? However, the dtmfmode should be the same, i think so, too, but...

Hi List, I'm having a problem with detecting incoming dtmf tones with chan_capi, using an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0, expecting that the capi module will detect the tones, but it did not. I also set both to 1, expecting that the asterisk dsp functions will detect them but it did not either. Can anyone provide any ideas how to overcome this issue? Esteban __________________________________________...