[Freeswitch-users] DTMF Issue

Thu Jun 5 10:36:54 MSD 2014

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> 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> 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
>> *Service Request Email: support at nysolutions.com <support at nysolutions.com>
>> *
>>
>> Polycom Certified VAR
>> Microsoft Small Business Specialist, Cisco SMB Select Certified
>>
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>>
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>>
>> *From:* 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> 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> 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> 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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