[asterisk-bugs] [Asterisk 0013209]: DTMF RFC2833 via SIP is not working

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Dec 5 17:20:27 CST 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13209 
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Reported By:                ip-rob
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   13209
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.21.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-07-31 08:23 CDT
Last Modified:              2008-12-05 17:20 CST
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Summary:                    DTMF RFC2833 via SIP is not working
Description: 
Using provider bandwidth.com which support RFC2833.  Configure outbound
trunk to use dtmfmode=rfc2833 and we receive double digits on a different
asterisk servers IVR.  American Express IVR does not accept any digits
(used main customer service line to test entering credit card number). 
Other IVR functions do not work.

Changing to inband works but inband should not be required by
bandwidth.com, they support rfc2833.

Configuration is using SIP devices and SIP trunks only.  A search in issue
tracker found similar problems in January of 2008 but no currently open
issues.
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---------------------------------------------------------------------- 
 (0095893) bujones (reporter) - 2008-12-05 17:20
 http://bugs.digium.com/view.php?id=13209#c95893 
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I did some testing today since I was told that having the same sequence
number for the END packets was a bad thing. I read RFC2833 and from my read
of it the sequence number of the last packet should not matter. So to prove
my point I changed rtp.c to increment the sequence number of the END packet
and got the same results.

I was also told that a packet gap of 20ms was not a good thing either.
Again according to RFC2833 a packet gap of up 50ms should not result in a
gap in the tone. So 20ms is within the range. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-05 17:20 bujones        Note Added: 0095893                          
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