[Freeswitch-users] DTMF missing

Fri Dec 24 13:40:13 MSK 2010

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 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..Via:
SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" <
sip:7028 at 192.168.2.190
<sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
<sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..Call-ID:
0017592a-eb33001a-
  63da3294-1a7bfdfa at 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 at 192.168.2.49:5060;transport=udp>..Proxy-Authorization: Digest
username="7028"
  ,realm="192.168.2.190",uri="sip:7050 at 192.168.2.190<sip%3A7050 at 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 at 192.168.2.190
<sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
<sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID:
0017592a-eb33001a-63da3
  294-1a7bfdfa at 192.168.2.49..CSeq: 102 INVITE..Contact:
<sip:7050 at 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 at 192.168.2.190<sip%3A7050 at 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 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..To: <
sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..From: 7001<
sip:7001 at 192.168.2.190 <sip%3A7001 at 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 at 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 at 192.168.2.190
<sip%3A7050 at 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 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..To: <
sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID:
d32ffe546570a77e..C
  Seq: 2 INVITE..Contact:
<sip:7050 at 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 at 192.168.2.190
<sip%3A7050 at 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
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