[Freeswitch-users] DTMF missing
Sam
u2nsam at gmail.com
Tue Dec 28 12:11:07 MSK 2010
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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