[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
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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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