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

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Aug 13 15:14:55 CDT 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:                
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Project:                    Asterisk
Issue ID:                   13209
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
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:              2008-08-13 15:14 CDT
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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.
====================================================================== 

---------------------------------------------------------------------- 
 (0091382) blitzrage (administrator) - 2008-08-13 15:14
 http://bugs.digium.com/view.php?id=13209#c91382 
---------------------------------------------------------------------- 
In order to debug this issue, we'll need a full trace of the call,
including the RTP stream since that is where the RFC2833 information will
reside. This is probably best provided through a wireshark trace. In
addition to that, could you provide the 'rtp debug' information from the
console so we can compare what Asterisk thinks it is sending, and what the
wireshark trace is actually showing happening?

Thanks!
Leif. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-13 15:14 blitzrage      Note Added: 0091382                          
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