[Freeswitch-users] DTMF Issue
Francis
sms at icefire.qza.net.au
Sat May 31 15:39:41 MSD 2014
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.
>
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> *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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