[Freeswitch-users] DTMF Problem with T-Mobile carrier

[Kristian Kielhofner]

Phil,

  Broadvox is known to do quite a few things that can affect DTMF
handling in non-straightforward ways.  This isn't a criticism of
Broadvox, it's just very common for carriers mixing upstreams
(especially when using Sonus).

  How far have you debugged this with Broadvox?  At minimum you'll
probably need a full SIP+RTP capture of the incoming call (especially
RFC2833 events) for anyone to help.

On Mon, Oct 3, 2011 at 10:33 AM, Phil Kim <phil.kim at valiant.com> wrote:
> Hello everybody,
>
>
>
> I really need some help here!
>
>
>
> I’m trying to use FS for making both inbound and outbound calls through
> Broadvox.
>
> Everything works perfectly except one particular situation with DTMF
> problem.
>
> In case of inbound calling (I’m calling FS), everything works fine. Call
> connects and I can send DTMF to FS and FS recognized those 100% (Testing
> with softphone from my desktop, SIP phone in the office, and all wireless
> carriers)
>
> However, I am having trouble with outbound calling. I have set it up to
> where FS makes an outbound call to certain number and once the call is
> answered, we start playing some custom IVR.
>
> Here, it works fine with softphone from my desktop, SIP phones in the
> office, AT&T, and Verizon. But for some reason FS will not recognize the
> DTMF coming from T-Mobile cell phones.
>
> Anyone has any idea? Below are my current setting from external.xml which
> was recommended by Broadvox.
>
>
>
> <param name="codec-prefs" value="PCMU,G729"/>
>
>             <param name="dtmf-type" value="rfc2833"/>
>
>             <param name="auto-rtp-bugs" value="clear"/>
>
>             <param name="pass-rfc2833" value="true"/>
>
>             <param name="inbound-late-negotiation" value="true"/>
>
>
>
>             <param name="sip-trace" value="no"/>
>
>             <param name="sip-capture" value="no"/>
>
>             <param name="rfc2833-pt" value="101"/>
>
>             <param name="sip-port" value="$${external_sip_port}"/>
>
>             <param name="dialplan" value="XML"/>
>
>             <param name="context" value="public"/>
>
>             <param name="dtmf-duration" value="1000"/>
>
>             <param name="inbound-codec-prefs"
> value="$${global_codec_prefs}"/>
>
>             <param name="outbound-codec-prefs"
> value="$${outbound_codec_prefs}"/>
>
>             <param name="hold-music" value="$${hold_music}"/>
>
>             <param name="rtp-timer-name" value="soft"/>
>
>
>
>
>
>
>
> Best regards,
>
>
>
> Phil Kim
>
>
>
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-- 
Kristian Kielhofner