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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