DTMF detection giving problems
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
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@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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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@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@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@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@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@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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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Narrowing down the problem, I see the PlayandGetDTMF starting the intended audio file. Then user presses any DTMF and the PAGD breaks saying the user pressed " " , yes and empty space :S.The variable which is supposed to hold the DTMF has some space in it which it proudly logs to the console w/o giving any errors !! On Fri, Aug 10, 2012 at 9:55 AM, SamyGo <govoiper@gmail.com> wrote:
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
You can see the script? 2012/8/10 SamyGo <govoiper@gmail.com>
Narrowing down the problem, I see the PlayandGetDTMF starting the intended audio file. Then user presses any DTMF and the PAGD breaks saying the user pressed " " , yes and empty space :S.The variable which is supposed to hold the DTMF has some space in it which it proudly logs to the console w/o giving any errors !!
On Fri, Aug 10, 2012 at 9:55 AM, SamyGo <govoiper@gmail.com> wrote:
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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participants (2)
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ivan baydin
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SamyGo