Everything works perfectly!

 

Thanks

 

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Monday, August 26, 2013 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Tuning DTMF

 

Yep that's perfectly normal. As long as FS is interpreting the DTMF correctly you are good to go. As a quick test if you are still unsure call into FS and transfer to the demo ivr and see if it does what you expect.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster@davri.com
Indianapolis, Indiana

Sent from a mobile device.

On Aug 26, 2013 2:32 AM, "Grant Bagdasarian" <GB@cm.nl> wrote:

Yes, that works fine. The cracking DTMF tones only happen when the remote party is sending DTMF tones. So if I call my cellphone with X-Lite through FS and my cellphones starts sending DTMF tones, that’s when the tones come in with some sort of cracking sound.

 

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Saturday, August 24, 2013 3:13 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Tuning DTMF

 

Have you tried using Xlite on an IVR or something? Call your bank or your cable company and see if you can get through the menus fine. I'd bet that you can. Post back and let us know. They sound short but that should be ok.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster@davri.com
Indianapolis, Indiana

Sent from a mobile device.

On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB@cm.nl> wrote:

Hello,

 

Setup is as following:

 

X-Lite (Test phone) ------>  SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone

 

Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone.

 

Console debug:

2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847

2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280 seq=5099 lw=161

2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280 seq=5100 lw=161

2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161

2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280 seq=5102 lw=161

2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161

2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161

2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1

2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1

2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1

2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280

2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280

 

Both internal and external sip profiles have the following configuration for DTMF:

<param name="liberal-dtmf" value="true"/>

<param name="rfc2833-pt" value="101"/>

<param name="dtmf-duration" value="2000"/>

<param name="inbound-codec-negotiation" value="generous"/>

<param name="inbound-late-negotiation" value="true"/>

 

Internal.xml (Dialplan)

    <extension name="internal_to_external">

      <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">

        <action application="set" data="hangup_after_bridge=false"/>

        <action application="set" data="continue_on_fail=true"/>

        <action application="bridge" data="sofia/external/${sip_req_uri}"/>

      </condition>

    </extension>

 

Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit?

 

Thanks!

 

Grant

 


_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org