[Freeswitch-users] DTMF detection giving problems

Fri Aug 10 12:56:32 MSD 2012

Did you terminate with the "#" DTMF char?

/Peter

Från: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] För SamyGo
Skickat: den 10 augusti 2012 10:38
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] DTMF detection giving problems

OK. Fixed that. Now, The thing is when I was testing it I never waited for the menu to end and just pressed buttons normally. And even then I couldn't get any DTMF.

On Fri, Aug 10, 2012 at 1:30 PM, SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>> wrote:
is that ms ? God bless the poor callers !! Goodness. wait I'll be back.

On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
7 ms seems like a very short timeout to have?

/Peter

Från: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] För SamyGo
Skickat: den 10 augusti 2012 10:11
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] DTMF detection giving problems

No Ivan , I've no issue at all with DTMFs ever. Like I said there is an Asterisk server which used torun the same IVRs and on same setup and never got such a problem.
But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in between. Calls are coming in OK, I intermittently face this issue where user presses DTMFs and those digits are not recognized by FS-PlayAndGetDigits()
I've  lines like this in my LUA

 choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "", "[0-9]")
 freeswitch.consoleLog("INFO","User Entered "..choice.." for this Survey\n")

Logs for the calls without dtmf here:

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280

--------------------- Another Example---------------------

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm

--------------------- Another Example---------------------

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm

Log for successful DTMF collected is as follows:

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]
[INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm
-------

So I can see a pattern here. But I can't figure out this behavior. Is it because I've 1 try only ! and after just once playing of file PAGD() breaks no matter it gets DTMF or not !?

Regards,
Sammy

On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com<mailto:runtwistedrat at gmail.com>> wrote:
Hello Sammy.
should check whether the operator is not blocking dtmf
I had a situation - in the same statement it worked, on the other - not processed incoming dtmf

2012/8/10 SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>>
Hello All,

I've a small lua script which is just an IVR. Calls are landing on Asterisk and then they are relayed over to FS. The problem is that the IVR don't always detects what user pressed. What I've observed is that the DTMF threshold is very high and some people just press slowly/quickly and the tone duration gets low and FS don't detect anything.

I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've also liberal-dtmf set in my sofia external.xml profile file which somewhat got me better results but still DTMFs miss sometimes.

Please note that the asterisk on which the calls are landing have never faced this issue before. Asterisk has PRI from which it receives the calls and then relay to desired FS server.

Regards.
Sammy

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