Hello

Just like Afshin, I am also missing DTMF digits ,using FreeSWITCH Version 1.0.head (git-34a0ca5 2010-12-22 20-38-57 -0600) ;
should i upgrade to latest ?

Regards
Sam



On Fri, Dec 24, 2010 at 4:10 PM, Sam <u2nsam@gmail.com> wrote:
Hi,,


I have installed the latest ver of freeswitch and i have configured the conference.

now when i punch in the digits for password , i could see that the DTMF digits are missed on fs_cli.

it only happens when i dial it from polycom or cisco phones.

I have tried with and without these values below:-

<param name="dtmf-type" value="rfc2833"/>
    <param name="pass-rfc2833" value="true"/>



traces fetched:
192.168.2.49:5060 -> 192.168.2.190:5060
  INVITE sip:7050@192.168.2.190 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" <sip:7028@192.168.2.190>;tag=0017592aeb3305185b4a37ba-615f498d..To: <sip:7050@192.168.2.190>..Call-ID: 0017592a-eb33001a-
  63da3294-1a7bfdfa@192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec 2010 10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact: <sip:7028@192.168.2.49:5060;transport=udp>..Proxy-Authorization: Digest username="7028"
  ,realm="192.168.2.190",uri="sip:7050@192.168.2.190",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires: 180..Accept: application/sdp
  ..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub..Content-Length: 220..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 16102
   0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8 18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv..

  
192.168.2.190:5060 -> 192.168.2.49:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" <sip:7028@192.168.2.190>;tag=0017592aeb3305185b4a37ba-615f498d..To: <sip:7050@192.168.2.190>;tag=2XXUZpgr1rvgc..Call-ID: 0017592a-eb33001a-63da3
  294-1a7bfdfa@192.168.2.49..CSeq: 102 INVITE..Contact: <sip:7050@192.168.2.190:5060;transport=udp>..User-Agent: NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIF
  Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Session-Expires: 180
  0;refresher=uas..Min-SE: 120..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 249..Remote-Party-ID: "7050" <sip:7050@192.168.2.190>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1293163579 129
  3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..

But the dtmf are not missed when punched on eyebeam softphone.

And all the phones have RFC 2833.

traces fetched for softphone:-

192.168.2.17:6182 -> 192.168.2.190:5060
  INVITE sip:7050@192.168.2.190 SIP/2.0..To: <sip:7050@192.168.2.190>..From: 7001<sip:7001@192.168.2.190>;tag=6c557c1e..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID: d32ffe546570a77e..CS
  eq: 2 INVITE..Contact: <sip:7001@192.168.2.17:6182>..Max-Forwards: 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest username="7001",rea
  lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri="sip:7050@192.168.2.190",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent: eyeBeam release 3007n
   stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN IP4 192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398 RTP/AVP 0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17 6398..a=fmtp:101 0-15..a=rtpmap:
  101 telephone-event/8000..a=sendrecv..
 
 
192.168.2.190:5060 -> 192.168.2.17:6182
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From: 7001 <sip:7001@192.168.2.190>;tag=6c557c1e..To: <sip:7050@192.168.2.190>;tag=H0ctQv7KNgU2j..Call-ID: d32ffe546570a77e..C
  Seq: 2 INVITE..Contact: <sip:7050@192.168.2.190:5060;transport=udp>..User-Agent: NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support
  ed: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Session-Expires: 1800;refresher=uas..Min-SE: 120..
  Content-Type: application/sdp..Content-Disposition: session..Content-Length: 249..Remote-Party-ID: "7050" <sip:7050@192.168.2.190>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1293161640 1293161641 IN IP4 192.168.2.190..
  s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..


Any thing you can think how it can happen?


Regards
Sam