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[Freeswitch-users] DTMF from cell phones


 
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jan.kubr at gmail.com
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PostPosted: Fri Dec 05, 2008 6:09 am    Post subject: [Freeswitch-users] DTMF from cell phones Reply with quote

Hi,
recently someone was mentioning an issue with DTMF here, but there was
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
(read app doesn't terminate). I could not find any regularity in this,
sometimes it is recognized just fine, sometimes I had to wait for the
file to be played etc. The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is
recognized perfectly. So it's probably related to GSM or something.

I was wondering whether anyone experienced the same and whether there
is something I can do about it. There are a few DTMF-related variables
in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
them a bit, but I don't really know what I'm doing.. Couldn't find any
docs, either.
Any ideas would be appreciated.

Jan Kubr

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ack at telefonica.net
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PostPosted: Fri Dec 05, 2008 7:39 am    Post subject: [Freeswitch-users] DTMF from cell phones Reply with quote

I had some issues with some previous versions of FS , in trunk looks
that is fixed. ( Notice current svn revision is 10609 )

in sip profiles i have :

...
<param name="rfc2833-pt" value="101"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${global_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="none"/>
<param name="inbound-codec-negotiation" value="greedy"/>
...

As codecs g711 ULAW (PCMU):

in vars.xml.conf :

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM"/>

So i guess that using latest version with a few changes in your config
should work unless there's any other issue related to your sip provider
( PSTN / Media Gateway ), on this case you can get some captures of
sip/rtp traffic to check SDP and rtp Marks.

El vie, 05-12-2008 a las 12:08 +0100, Jan Kubr escribió:
Quote:
Hi,
recently someone was mentioning an issue with DTMF here, but there was
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
(read app doesn't terminate). I could not find any regularity in this,
sometimes it is recognized just fine, sometimes I had to wait for the
file to be played etc. The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is
recognized perfectly. So it's probably related to GSM or something.

I was wondering whether anyone experienced the same and whether there
is something I can do about it. There are a few DTMF-related variables
in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
them a bit, but I don't really know what I'm doing.. Couldn't find any
docs, either.
Any ideas would be appreciated.

Jan Kubr


Cheers,
--
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD 6BCF AC2C CA61 6EF1 B90D


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mike at jerris.com
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PostPosted: Fri Dec 05, 2008 9:55 am    Post subject: [Freeswitch-users] DTMF from cell phones Reply with quote

On Dec 5, 2008, at 6:08 AM, Jan Kubr wrote:

Quote:
Hi,
recently someone was mentioning an issue with DTMF here, but there was
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
(read app doesn't terminate). I could not find any regularity in this,
sometimes it is recognized just fine, sometimes I had to wait for the
file to be played etc. The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is
recognized perfectly. So it's probably related to GSM or something.

I was wondering whether anyone experienced the same and whether there
is something I can do about it. There are a few DTMF-related variables
in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
them a bit, but I don't really know what I'm doing.. Couldn't find any
docs, either.
Any ideas would be appreciated.

If it is coming from the sip provider as rfc 2833 dtmf, they are
probably doing inband detection and failing at it. If you look at an
rtp dump you can confirm this. If this is the case, there is nothing
you can do on the FreeSWITCH side and the provider will have to fix it.

Mike

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jan.kubr at gmail.com
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PostPosted: Fri Dec 05, 2008 1:41 pm    Post subject: [Freeswitch-users] DTMF from cell phones Reply with quote

Quote:
Quote:
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is


Quote:
So i guess that using latest version with a few changes in your config
should work unless there's any other issue related to your sip provider
( PSTN / Media Gateway ), on this case you can get some captures of
sip/rtp traffic to check SDP and rtp Marks.

I tried trunk and the values for the variables (all except
rtp-timer-name=none are already default in trunk), but only two things
are different:
1. When I press a key, the read app seem to always terminate, but not
always the dtmf is captured in a variable.
2. The read app seems to ignore the variable name parameter: calling
it with "1 1 104.wav choice_181152 10000 #" doesn't put the digit in
variable_choice_181152, but to dmtf_digit. Why is that?

Quote:
If it is coming from the sip provider as rfc 2833 dtmf, they are
probably doing inband detection and failing at it. If you look at an
rtp dump you can confirm this. If this is the case, there is nothing
you can do on the FreeSWITCH side and the provider will have to fix it.

But the call goes through the same SIP provider even when using the
soft phone and there it works fine. The difference might be that then
it is SIP to SIP within the same provider.. How do I do the RTP dump?

Also I should have mentioned that DTMF is not captured only DURING the
file is being played. It is always captured correctly when I wait
until the playback is finished. Does this sound familiar? I thought
this would be somet obvious misconfiguration on my side.


Jan

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