[Freeswitch-users] DTMF detection giving problems

[SamyGo]

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>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>
>
>>  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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>
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