[asterisk-bugs] [Asterisk 0016292]: [patch] DTMF Not Recognized with Exchange UM

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16292 
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Reported By:                rsw686
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16292
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     confirmed
Asterisk Version:           1.6.1.10 
JIRA:                       SWP-463 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-11-20 09:53 CST
Last Modified:              2009-12-03 20:57 CST
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Summary:                    [patch] DTMF Not Recognized with Exchange UM
Description: 
I just upgraded my Asterisk 1.6.1.4 version to 1.6.1.10 and the DTMF tones
are not being recognized by the Exchange UM server. I tested out the RC
release and found that it worked in 1.6.1.7-rc1 and broke in 1.6.1.7-rc2.

What happens is Exchange UM will say to enter your mailbox number. I enter
8532 and it will say that 8 is not a valid entry. It appears Asterisk is
only detecting the first DTMF tone.
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 (0114704) rsw686 (reporter) - 2009-12-03 20:57
 https://issues.asterisk.org/view.php?id=16292#c114704 
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I've added two more patches (asterisk-1.6.1.11-constantssrc-debug.patch and
asterisk-1.6.1.11-constantssrc.patch). The debug has verbose logging so you
can see what is going on and the other is my final patch for the constant
ssrc dtmf issue. The final patch applies against 1.6.2 as well.

Here is the console output with the debug patch. Notice that the
constantssrc flag is being properly set after forwarding the SIP channel
with the lines I added to sip_pvt. Also note the redundant setting of the
flag in handle_request_invite, which I think should being removed. Both
sip_pvt and create_addr_from_peer follow the same structure and set the
dtmf and rtp timeouts for the audio and video streams along with the
constantssrc flag. The handle_request_invite constantssrc flag setting
seems out of place.

  == Using SIP RTP CoS mark 5
sip_pvt: setting constantsrc flag for audio
handle_request_invite: setting constantsrc flag for audio
    -- Executing [3300 at default:1] Answer("SIP/38678-00000000", "") in new
stack
ast_rtp_new_source: constantssrc is 1, ssrc is 476670726
    -- Executing [3300 at default:2] Dial("SIP/38678-00000000",
"SIP/SIP_VM/3300,,TTr") in new stack
  == Using SIP RTP CoS mark 5
sip_pvt: setting constantsrc flag for audio
create_addr_from_peer: setting constantsrc flag for audio
    -- Called SIP_VM/3300
    -- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13
    -- Now forwarding SIP/38678-00000000 to
'SIP/3300::::TCP at 10.9.1.13:5067' (thanks to SIP/SIP_VM-00000001)
  == Using SIP RTP CoS mark 5
sip_pvt: setting constantsrc flag for audio
    -- SIP/10.9.1.13:5067-00000002 is ringing
    -- SIP/10.9.1.13:5067-00000002 answered SIP/38678-00000000
ast_rtp_new_source: constantssrc is 1, ssrc is 476670726
ast_rtp_new_source: constantssrc is 1, ssrc is 989379290
ast_rtp_new_source: constantssrc is 1, ssrc is 476670726
ast_rtp_new_source: constantssrc is 1, ssrc is 989379290 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-03 20:57 rsw686         Note Added: 0114704                          
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