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Summary:ASTERISK-21522: [patch] DTMF end is not always processed, causes one-way audio
Reporter:Corey Farrell (coreyfarrell)Labels:
Date Opened:2013-04-17 17:05:58Date Closed:2013-05-01 16:26:38
Priority:MajorRegression?
Status:Closed/CompleteComponents:Resources/res_rtp_asterisk
Versions:SVN 1.8.22.0 11.4.0 Frequency of
Occurrence
Frequent
Related
Issues:
Environment:Attachments:( 0) rtp_dtmf_process_end.patch
Description:Some Cisco IVR's will tell Asterisk to stop sending RTP as soon as DTMF start is received.  Then the Cisco redirects the call to a new destination.  Once the IVR destination answers, asterisk resumes sending DTMF since we were not sending rtp when it should have processed DTMF end.  This causes one-way audio for the duration of the call.

My patch has been tested on the customer's system with Asterisk 11.2.1.
Comments:By: Corey Farrell (coreyfarrell) 2013-04-17 17:07:28.511-0500

Ensure rtp->sending_digit and rtp->send_digit are always zero after DTMF end.