[asterisk-bugs] [Asterisk 0017372]: Progress in band error (don't send RTP packets)
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jun 3 12:37:46 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=17372
======================================================================
Reported By: tech_admin
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 17372
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.6.2.7
JIRA: SWP-1526
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-05-21 09:31 CDT
Last Modified: 2010-06-03 12:37 CDT
======================================================================
Summary: Progress in band error (don't send RTP packets)
Description:
Dear all,
on a updated redhat rhel5 (kernel: 2.6.18-194.3.1.el5xen), I installed the
last asterisk release (1.6.2.7) from the source code.
The problem is: SIP caller doesn't hear ringing tone.
We use progressinband=yes with ch (Switzerland) tones and voice is G729
encoded using TC400B Digium card.
After answer, all is ok: both side RTP packets go trough asterisk without
translation.
The Digium card work fine with others service like IVR, Queues, Voicemail
... all using G729.
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0017185 [patch] [regression] Using Local channe...
======================================================================
----------------------------------------------------------------------
(0122908) DennisD (reporter) - 2010-06-03 12:37
https://issues.asterisk.org/view.php?id=17372#c122908
----------------------------------------------------------------------
All of my attached debugs/captures were run with:
exten => 520,1,NoOp()
exten => 520,n,Progress()
exten => 520,n,Wait(1)
exten => 520,n,Proceeding()
exten => 520,n,Wait(1)
exten => 520,n,Ringing()
exten => 520,n,Wait(60)
exten => 520,n,Hangup()
When testing with Future-Nine and voip.ms, the first statement was
Progress() in the dialplan.
Issue History
Date Modified Username Field Change
======================================================================
2010-06-03 12:37 DennisD Note Added: 0122908
======================================================================
More information about the asterisk-bugs
mailing list