[Freeswitch-users] uuid_send_dtmf problem with rfc2833

[Michael Jerris]

maybe nat issue?  It's impossible to guess with this little information.


> On Nov 24, 2014, at 6:56 AM, Ítalo Rossi <italorossib at gmail.com> wrote:
> 
> I also had this problem, a workaround is send any audio before sending the dtmf, it'll work.
> 
> Does anyone know why this happen?
> 
> On Sat, Nov 22, 2014 at 7:42 PM, Adam Kuśmirek <amkusmirek at gmail.com <mailto:amkusmirek at gmail.com>> wrote:
> Hello,
> 
> I have a problem with sending dtmf events (rfc2833) with uuid_send_dtmf api command.
> 
> FreeSWITCH Version 1.5.15b+git~20141118T231404Z~df423b88d6~64bit (git df423b8 2014-11-18 23:14:04Z 64bit)
> 
> First i originate the call and park it with the command:
> 
> originate {origination_caller_id_name=test}sofia/gateway/test/xxxxxxxx at yyy.zzz.com <mailto:xxxxxxxx at yyy.zzz.com> &park()
> 
> The other side answers the call. I can see that rfc 2833 is negotiated in SDPs.
> 
>    SIP/2.0 200 Ok
>    Via: SIP/2.0/UDP 46.174.232.238:5070;rport;branch=z9hG4bKFmQvapcNNp44m
>    From: "..." <sip:... at ...>;tag=35FN6HXZ3ry2F
>    To: <sip:12122777254 at sip.12voip.com <mailto:sip%3A12122777254 at sip.12voip.com>>;tag=f0313ac544a16a82aee0b
>    Contact: sip:... at ...:5060
>    Call-ID: f7b0d42d-ed32-1232-06a5-c03fd5651f3e
>    CSeq: 68010140 INVITE
>    Server: (Very nice Sip Registrar/Proxy Server)
>    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
>    Content-Type: application/sdp
>    Content-Length: 211
> 
>    v=0
>    o=xxx 1416695524 1416695524 IN IP4 yyyy
>    s=SIP Call
>    c=IN IP4 yyyy
>    t=0 0
>    m=audio 27792 RTP/AVP 8 101
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
> 
> 
> Then I try to send DTMF
> 
> freeswitch at internal> uuid_send_dtmf 6022b884-7297-11e4-8682-fd8246740d0e 12345
> -ERR no reply
> 
> 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com <mailto:12122777254 at sip.12voip.com> send dtmf
> digit=1 ms=250 samples=2000
> 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com <mailto:12122777254 at sip.12voip.com> send dtmf
> digit=2 ms=250 samples=2000
> 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com <mailto:12122777254 at sip.12voip.com> send dtmf
> digit=3 ms=250 samples=2000
> 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com <mailto:12122777254 at sip.12voip.com> send dtmf
> digit=4 ms=250 samples=2000
> 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com <mailto:12122777254 at sip.12voip.com> send dtmf
> digit=5 ms=250 samples=2000
> 
> 
> Unfortunatelly it doesn't work and I can't see any rtpevents in wireshark.
> 
> If I change dtmf-type in profile to info, i can see  sip info packets.
> 
> 
> Please help
> 
> 

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