[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:                     ready for testing
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:              2009-12-01 13:25 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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---------------------------------------------------------------------- 
 (0114472) rsw686 (reporter) - 2009-12-01 13:25
 https://issues.asterisk.org/view.php?id=16292#c114472 
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I added some verbose logging to main/rtp.c.

void ast_rtp_set_constantssrc(struct ast_rtp *rtp)
{
        ast_verbose("ast_rtp_set_constantssrc: setting flag\n");
        rtp->constantssrc = 1;
}

void ast_rtp_new_source(struct ast_rtp *rtp)
{
        if (rtp) {
                rtp->set_marker_bit = 1;
                ast_verbose("ast_rtp_new_source: constantssrc is %d\n",
rtp->constantssrc);
                if (!rtp->constantssrc) {
                        rtp->ssrc = ast_random();
                }
        }
}

Have a look at the asterisk output. The flag is getting set correctly, but
for some reason it is toggling back and forth when I press keys on the
phone. Shouldn't this stay set the entire call? If I comment out the
ast_random line in ast_rtp_new_source the DTMF tones work fine.

  == Using SIP RTP CoS mark 5
ast_rtp_set_constantssrc: setting flag
    -- Executing [3300 at default:1] Answer("SIP/38678-00000003", "") in new
stack
ast_rtp_new_source: constantssrc is 1
    -- Executing [3300 at default:2] Dial("SIP/38678-00000003",
"SIP/SIP_VM/3300,,TTr") in new stack
  == Using SIP RTP CoS mark 5
ast_rtp_set_constantssrc: setting flag
    -- Called SIP_VM/3300
    -- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13
    -- Now forwarding SIP/38678-00000003 to
'SIP/3300::::TCP at 10.9.1.13:5067' (thanks to SIP/SIP_VM-00000004)
  == Using SIP RTP CoS mark 5
    -- SIP/10.9.1.13:5067-00000005 is ringing
    -- SIP/10.9.1.13:5067-00000005 answered SIP/38678-00000003
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0
ast_rtp_new_source: constantssrc is 1
ast_rtp_new_source: constantssrc is 0 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-01 13:25 rsw686         Note Added: 0114472                          
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