[Freeswitch-users] DTMF missing
Sam
u2nsam at gmail.com
Mon Dec 27 10:56:23 MSK 2010
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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