[Freeswitch-users] DTMF missing
Sam
u2nsam at gmail.com
Wed Dec 29 07:29:32 MSK 2010
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 at 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 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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>
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