The client is sending DTMF inband

also have used <variable name="bypass_media" value="0"/>

and also have used

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

for IVR i could see the dtmf ; but when after bridging extension to extension i do not see the dtmf when used for attn transfer.

Regards
Sam



On Wed, Dec 29, 2010 at 3:21 AM, Michael Collins <msc@freeswitch.org> wrote:
Assuming your console in on DEBUG level output then yes, you should see the DTMFs show up on the screen. If you're not seeing DTMFs then yeah, something is wrong. Confirm whether the client is sending the DTMFs inband or not.

-MC


On Tue, Dec 28, 2010 at 1:11 AM, Sam <u2nsam@gmail.com> wrote:
When i do " #1 " i cannot see the dtmf on the fs_cli , is it because of that ?

Regds
Sam


On Mon, Dec 27, 2010 at 1:26 PM, Sam <u2nsam@gmail.com> wrote:
Hi ,

I have now upgraded to ver (git-4e95227 2010-12-26 09-09-14 -0600) , now the dtmfs in IVR are working !


When i used bind meta application then its not executing, any comments for me ?

            <action application="set" data="export_vars=#,*"/>
            <action application="export" data="#=true"/>
            <action application="set" data="bind_meta_key=#"/>
            <action application="bind_meta_app" data="1 ab s execute_extension::dx XML features"/>
           



Regards
Sam




On Sun, Dec 26, 2010 at 11:16 PM, afshin afzali <a.afzali2003@gmail.com> wrote:
Sam,

Yes, The issue has resolved in last git :)
-- afshin

On Sun, Dec 26, 2010 at 7:33 AM, Sam <u2nsam@gmail.com> wrote:
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




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