[Freeswitch-users] DTMF Issue

Francis sms at icefire.qza.net.au
Thu Jun 5 10:55:54 MSD 2014


Hi Richard, there is a good write up on DTMF here, which goes into the 
default behavior:

https://wiki.freeswitch.org/wiki/DTMF

As to why rfc2833 is misbehaving, as Brian said, we'll need to see a SIP 
trace, either from the console or from wireshark before any definitive 
cause can be found.

-Francis


On 5/06/2014 4:36 PM, Richard Mace wrote:
>
> Hi Francis,
> When I changed it to sip info in Zoiper it worked, but what I don't 
> understand is why it did work because it is commented out in 
> internal.XML and so therefore I thought that the setting would default 
> to rfc2833?
> Richard
>
> On 31 May 2014 12:48, "Francis" <sms at icefire.qza.net.au 
> <mailto:sms at icefire.qza.net.au>> wrote:
>
>     Hi Richard, the vanilla config uses RFC2833 by default, it doesn't
>     need to be set in the profile or the user account. Have you tried
>     any other clients, like Zoiper or Xlite? I've deployed Zoiper for
>     some of the users I support and once you set DTMF style (Accounts
>     --> Features), it works as expected.
>
>     Try also setting your codec to G711 (either variant) or GSM. These
>     are two of the default FS codecs and should work universally.
>
>     -Francis
>
>
>     On 30/05/2014 3:59 PM, Richard Mace wrote:
>>     Hi,
>>     If I put  <param name="dtmf-type" value="rfc2833"/> in SIP
>>     internal and set Jiti up to use RFC2833 then the DTMF gets
>>     ignored, however, if I change both to sip info, FreeSWITCH
>>     responds with:
>>
>>     IGNORE INFO DTMF(2) (This channel was not configured to use INFO
>>     DTMF!
>>
>>     Any ideas?
>>
>>     Also, this happens the same whether a phone is remote or onsite.
>>     Should the DTMF setting be put in sip internal, or should it be
>>     per end point in the directory?
>>
>>     Thanks
>>
>>     Richard
>>
>>
>>
>>     On 28 May 2014 13:43, Moishe Grunstein <max at nysolutions.com
>>     <mailto:max at nysolutions.com>> wrote:
>>
>>         Which codec are you using, which dtmf method? Do you see the
>>         dtmf's in a sip capture?
>>
>>         Thanks,
>>
>>         Moishe Grunstein
>>
>>         Tornado Computer Systems, Inc.
>>
>>         212.400.7650 888.IPPBX.US <http://888.IPPBX.US>
>>         *Service Request Email: support at nysolutions.com
>>         <mailto:support at nysolutions.com> *
>>
>>         Polycom Certified VAR
>>         Microsoft Small Business Specialist, Cisco SMB Select Certified
>>
>>         cid:image001.jpg at 01C72F94.9EE45D60 <http://www.nysolutions.com/>
>>
>>         Computer Networking * Managed Services * IP Video
>>         Surveillance * Network Assessments * Web Solutions * Voice
>>         over IP * Disaster Recovery * Network Security * Site Surveys
>>         * CMS
>>
>>         *From:*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>] *On
>>         Behalf Of *Richard Mace
>>         *Sent:* Wednesday, May 28, 2014 7:47 AM
>>         *To:* FreeSWITCH Users Help
>>         *Subject:* Re: [Freeswitch-users] DTMF Issue
>>
>>         Hi Francis,
>>
>>         Thanks for your reply.
>>
>>         On 27 May 2014 20:47, Francis <sms at icefire.qza.net.au
>>         <mailto:sms at icefire.qza.net.au>> wrote:
>>
>>         It just dawned on me (sleep deprivation catching up), 5060 to
>>         5070/10000 to 20000 is Asterisk, FS uses 5060 to 5081/16385
>>         to 32768. Try that on your router before reading anything
>>         below :)
>>
>>         Yes, I had my Asterisk hat on. Chnaged the ports to what you
>>         suggested, but it was still the same.
>>
>>             If that doesn't work, does the call drop after 30 seconds
>>             or so? If yes, you have a NAT issue. Either way, there's
>>             a few steps to try:
>>
>>         No, it doesn't cut off at all.
>>
>>             1. Make sure STUN is enabled on Jitsi
>>
>>         I had a look but couldn't find a STUN setting anywhere.?
>>
>>             2. Connect a second client to see if you can get audio
>>             both ways
>>
>>             3. Manually set external_sip_ip and external_rtp_ip in
>>             your SIP profile
>>
>>             4. Set aggressive_nat_detection to yes in your SIP profile
>>
>>         I will try these other suggestions later.
>>         Thanks again
>>
>>         Richard
>>
>>
>>             Step 4 will get things working in most cases, but it can
>>             break endpoints in certain circumstances. Use with
>>             caution. Some bedtime reading:
>>
>>             http://wiki.freeswitch.org/wiki/NAT
>>
>>             https://wiki.freeswitch.org/wiki/Auto_NAT
>>
>>             https://wiki.freeswitch.org/wiki/NAT_Traversal
>>
>>             https://wiki.freeswitch.org/wiki/General_NAT_example_scenarios
>>
>>             -Francis
>>
>>             On 2014-05-28 05:06, Richard Mace wrote:
>>
>>                 Hi Francis,
>>                 Thanks for your reply.
>>                 It was the actual demo ivr that I was trying because
>>                 I wanted to do an echo test by pressing the digit.
>>                 I can hear the DTMF dial but nothing happens?
>>
>>                 Richard
>>
>>                 On 27 May 2014 15:09, "Francis"
>>                 <sms at icefire.qza.net.au
>>                 <mailto:sms at icefire.qza.net.au>> wrote:
>>
>>                 Hi Richard,
>>
>>                 When you say "dial a DTMF digit", do you mean during
>>                 the call? If that's the case, you'll only get a
>>                 response from an in-call macro or IVR if the call is
>>                 local. Depending on the soft client, you may not hear
>>                 digits if you press them during a call, but they will
>>                 still go through. Try the demo IVR and see what happens.
>>
>>                 To your second question, if you are using RFC-2833
>>                 (the preferred method), then DTMF will be sent on the
>>                 RTP port, so you won't need any extra settings. If
>>                 you can send and receive audio, then DTMF should work.
>>
>>                 Francis
>>
>>                 On 27/05/2014 10:53 PM, Richard Mace wrote:
>>
>>                     Hi,
>>
>>                     Sorry to add to my original post, but, I have
>>                     opened ports 5060-5070 and 10000-20000.
>>
>>                     Is there a particular port that relates to DTMF
>>                     or is is likely to be some kind of NAT issue?
>>
>>                     Richard
>>
>>                     On 27 May 2014 07:31, Richard Mace
>>                     <richard.mace at gmail.com
>>                     <mailto:richard.mace at gmail.com>> wrote:
>>
>>                     Hi All,
>>
>>                     I have a FreeSWITCH server running behind a
>>                     firewall and can successfully connect a Jitsi
>>                     software client to the server remotely. I can
>>                     dial 5000 from Jitsi and hear audio, but when I
>>                     dial a DTMF digit,nothing happens.
>>
>>                     I have tried configuring Jitsi to use different
>>                     DTMF methods, but none seem to work.
>>
>>                     It is a vanilla install of FreeSWITCH, Running on
>>                     Debian 7 installed via the packages so the config
>>                     files will be default.
>>
>>                     Any ideas?
>>
>>                     Thanks
>>
>>                     Richard
>>
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