[asterisk-bugs] [Asterisk 0014164]: Dial() option d is not working

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Feb 6 11:19:22 CST 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14164 
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Reported By:                DennisD
Assigned To:                putnopvut
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Project:                    Asterisk
Issue ID:                   14164
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Target Version:             1.6.0.5
Asterisk Version:           1.6.0.3-rc1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 167059 
Request Review:              
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Date Submitted:             2009-01-02 15:08 CST
Last Modified:              2009-02-06 11:19 CST
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Summary:                    Dial() option d is not working
Description: 
"-- User hit 1 to disconnect call." doesn't show up and nothing happens
when using just the option d and pressing 1 when the call is ringing.

If I change the options to md, then it works (after a delay).  Also, if I
use retrydial, after it retries, then it works perfectly.

I have tried this with asterisk-1.6.0.3-rc1 and asterisk-1.6.0 from
SVN-branch-1.6.0-r167059M and they both do the same thing.


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---------------------------------------------------------------------- 
 (0099621) putnopvut (administrator) - 2009-02-06 11:19
 http://bugs.digium.com/view.php?id=14164#c99621 
---------------------------------------------------------------------- 
I'm just updating this bug to say that it's not being ignored. It's a
situation where I know why the problem is occurring. It's exactly as I
stated in my first note. There's no RTP session set up, so DTMF isn't being
sent or received properly. The problem is that I really don't have a good
idea of a way to go about solving this.

What I'm thinking is that since DTMF is "early media" in this scenario, we
need to send a 183 Session Progress with SDP if an incoming SIP call
attempts to Dial using the 'd' option (or if any other options allow for
pre-bridge DTMF presses). I'm not really sure of all the implications of
such an action, though, so I'm hesitant to just blindly add such a
concession without exploring its effects and investigating alternatives. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-06 11:19 putnopvut      Note Added: 0099621                          
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