[asterisk-bugs] [Asterisk 0014093]: PlayDTMF is not working

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Dec 16 13:52:07 CST 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14093 
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Reported By:                msetim
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14093
Category:                   Core/ManagerInterface
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     feedback
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:             2008-12-16 13:23 CST
Last Modified:              2008-12-16 13:52 CST
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Summary:                    PlayDTMF is not working
Description: 
Hi guys,

I'm trying to play dtmf digits to a channel, but it's not working. To test
its is very simple, I made:

0. Enable full in logger.conf (exec tail -f /var/log/asterisk/full)
1. Register two sip phones
2. Call from one to another
3. Get some channel (show channels)
4. Login manager (telnet localhost 5038)
5. Send event: 
Action: PlayDTMF
Channel: SIP/556-0859cdd8
Digit: 1 

Thanks guys.. I hope that additional information help to found root
cause.

Ps. I don't know the correct category and I'm receving an error when I try
to select .I did not set the category correctly. 


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---------------------------------------------------------------------- 
 (0096469) msetim (reporter) - 2008-12-16 13:52
 http://bugs.digium.com/view.php?id=14093#c96469 
---------------------------------------------------------------------- 
Hi file,

I tried to sent DTMF to both channels trying to figure out some "my own"
mistake, but it don't see DTMF an anyone of them. I will try to explain
what I'm expecting:

I made a call from SIP/115 to SIP/556 and I try to send "DTMF digit 1" to
SIP/115.

Yes, I can sent to you an rtp debug. Do you need an rtp debug from tcpdump
or can be from asterisk cli command (rtp debug). From tcpdump can you have
show me the command that you need?

I'm using RFC2833 (DTMFmode: rfc2833) in both peers (SIP/115 and SIP/556)

Bye 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-16 13:52 msetim         Note Added: 0096469                          
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