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@icefire.qza.net.au <mailto:sms@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@nysolutions.com <mailto:max@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@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@icefire.qza.net.au <mailto:sms@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@icefire.qza.net.au <mailto:sms@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@gmail.com <mailto:richard.mace@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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