[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 10:50:14 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16604
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Reported By: frawd
Assigned To:
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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:
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Date Submitted: 2010-01-14 09:28 CST
Last Modified: 2010-04-23 10:50 CDT
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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
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(0120829) frawd (reporter) - 2010-04-23 10:50
https://issues.asterisk.org/view.php?id=16604#c120829
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Agreed for it being a dahdi issue, as it works ok in SIP to SIP or SIP to
DAHDI-PRI situations.
You can close this one and refer to
https://issues.asterisk.org/view.php?id=17224
Issue History
Date Modified Username Field Change
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2010-04-23 10:50 frawd Note Added: 0120829
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