[Freeswitch-users] DTMF missing
Michael Collins
msc at freeswitch.org
Wed Dec 29 00:51:43 MSK 2010
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 at 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 at 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 at gmail.com>wrote:
>>
>>> Sam,
>>>
>>> Yes, The issue has resolved in last git :)
>>> -- afshin
>>>
>>> On Sun, Dec 26, 2010 at 7:33 AM, Sam <u2nsam at 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 at 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 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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>>>>
>>>
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>>
>
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