[asterisk-bugs] [Asterisk 0016604]: Asterisk does not send "183 Session Progress" when dialing through a dahdi analog line
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Apr 23 11:06:41 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=16604
======================================================================
Reported By: frawd
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 16604
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: acknowledged
Asterisk Version: 1.6.2.1-rc1
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-01-14 09:28 CST
Last Modified: 2010-04-23 11:06 CDT
======================================================================
Summary: Asterisk does not send "183 Session Progress" when
dialing through a dahdi analog line
Description:
Scenario is a simple bridge between a SIP phone and an FXO line with
polarityswitch options turned on (and working). While the PSTN destination
hasn't answered yet, the SIP phone keeps silent and does not generate a
ring tone.
In 1.4 versions of Asterisk, after dialing the digits in the FXO, Asterisk
was sending a "183 Session Progress" back to the phone so it could generate
a progress ring.
In 1.6.2.1-rc1, it does not send it, probably related to some strange
failure to read frames:
[Jan 14 16:13:00] DEBUG[10593]: audiohook.c:248 audiohook_read_frame_both:
Failed to get 160 samples from read factory 0xa71d70
======================================================================
----------------------------------------------------------------------
(0120834) frawd (reporter) - 2010-04-23 11:06
https://issues.asterisk.org/view.php?id=16604#c120834
----------------------------------------------------------------------
Ok, after reading the UPGRADE.txt, it appears this is a total non-bug as
you mentioned.
It works perfectly by adding a Progress() before my Dial command.
This bug can really be closed.
Issue History
Date Modified Username Field Change
======================================================================
2010-04-23 11:06 frawd Note Added: 0120834
======================================================================
More information about the asterisk-bugs
mailing list