[asterisk-bugs] [Asterisk 0018084]: DTMF tones fail DAHDI>DAHDI

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Oct 7 08:16:28 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18084 
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Reported By:                seandarcy
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18084
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.2.14-rc1 
JIRA:                       SWP-2331 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-10-01 12:54 CDT
Last Modified:              2010-10-07 08:16 CDT
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Summary:                    DTMF tones fail DAHDI>DAHDI
Description: 
I have internal DAHDI lines and an external DAHDI PSTN for local calls and
use SIP for ld calls. If I call locally (DAHDI > DAHDI) the dtmf tones are
messed up. They seem to be doubled, that is a "1" becomes "11". 

But on ld calls (DAHDI > SIP) dtmf works! And I tried calling a local
number that didn't work over PSTN.



======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0017066 Adaptive Jitter Buffer issue
====================================================================== 

---------------------------------------------------------------------- 
 (0127778) aragon (reporter) - 2010-10-07 08:16
 https://issues.asterisk.org/view.php?id=18084#c127778 
---------------------------------------------------------------------- 
I don't understand the relationship to
https://issues.asterisk.org/view.php?id=17066 either.
I am also able to reproduce this problem and jitter buffer is disabled.
Technology is PRI.
Incoming channel forwarded to outgoing channel (bridged PRI B channels).
Call terminates on banking IVR and all DTMF is double digited making IVR
usage impossible on forwarded call.
libpri 1.4.11
zaptel 1.4.12
Asterisk 1.4.35

Typical DTMF CLI output when caller dials 0 on IVR

[Apr 28 09:47:16] DTMF[8091]: channel.c:2387 __ast_read: DTMF begin
'0'received on Zap/8-1
[Apr 28 09:47:16] DTMF[8091]: channel.c:2397 __ast_read: DTMF begin
passthrough '0' on Zap/8-1
[Apr 28 09:47:16] DTMF[8091]: channel.c:2318 __ast_read: DTMF end '0'
received on Zap/8-1, duration 80 ms
[Apr 28 09:47:16] DTMF[8091]: channel.c:2355 __ast_read: DTMF end accepted
with begin '0' on Zap/8-1
[Apr 28 09:47:16] DTMF[8091]: channel.c:2371 __ast_read: DTMF end
passthrough '0' on Zap/8-1
[Apr 28 09:47:16] DTMF[8093]: channel.c:2387 __ast_read: DTMF begin
'0'received on Local/94165551234 at tenant-outgoing-b99d,2
[Apr 28 09:47:16] DTMF[8093]: channel.c:2397 __ast_read: DTMF begin
passthrough '0' on Local/94165551234 at tenant-outgoing-b99d,2
[Apr 28 09:47:16] DTMF[8093]: channel.c:2318 __ast_read: DTMF end '0'
received on Local/94165551234 at tenant-outgoing-b99d,2, duration 100 ms
[Apr 28 09:47:16] DTMF[8093]: channel.c:2355 __ast_read: DTMF end accepted
with begin '0' on Local/94162014196 at tenant-outgoing-b99d,2
[Apr 28 09:47:16] DTMF[8093]: channel.c:2371 __ast_read: DTMF end
passthrough '0

I have been watching progress of https://reviewboard.asterisk.org/r/625/
wondering if this might help solve this problem.  But this review may be
100% unrelated. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-10-07 08:16 aragon         Note Added: 0127778                          
======================================================================




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