[asterisk-bugs] [Asterisk 0016795]: 1.4 does not send any SIP messages after the "100 Trying" to the T.38 INVITE requesting side
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jun 1 09:54:06 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16795
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Reported By: vrban
Assigned To:
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Project: Asterisk
Issue ID: 16795
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: minor
Priority: normal
Status: confirmed
Asterisk Version: 1.4.29
JIRA: SWP-891
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: 2010-02-09 11:07 CST
Last Modified: 2010-06-01 09:54 CDT
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Summary: 1.4 does not send any SIP messages after the "100
Trying" to the T.38 INVITE requesting side
Description:
Asterisk 1.4 does not send any SIP messages after the "100 Trying" to the
T.38 INVITE requesting side. chan_sip talk to the B-side, where itself send
out the T.38 re-INVITE, but does not send anything to the originating
A-side.
"sip show channels" show this dead A side channels which never get closed:
111.222.111.222 0123456789 69570b5c142b447 0x0 (nothing) No
Tx: INVITE
111.222.111.222 0123456789 03f0bc4832e47e8 0x0 (nothing) No
Tx: INVITE
111.222.111.222 0123456789 10a623dc4650f90 0x0 (nothing) No
Tx: INVITE
111.222.111.222 0123456789 5636aa6171efef3 0x0 (nothing) No
Tx: INVITE
111.222.111.222 0123456789 5877536469f446e 0x0 (nothing) No
Tx: INVITE
111.222.111.222 0123456789 3a83a6480cf7eed 0x0 (nothing) No
Tx: INVITE
111.222.111.222 0123456789 664d8ac26547274 0x0 (nothing) No
Tx: INVITE
111.222.111.222 0123456789 38bb8ada68b44ae 0x0 (nothing) No
Tx: INVITE
With 1.6 and trunk chan_sip is talking to A-side.
In the attached sip trace, you see the only answer to C.C.C.C is only the
100 Trying, then nothing.
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Relationships ID Summary
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related to 0016692 [patch] Missing fallback to audio fax f...
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(0122687) svnbot (reporter) - 2010-06-01 09:54
https://issues.asterisk.org/view.php?id=16795#c122687
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Repository: asterisk
Revision: 266579
U branches/1.4/channels/chan_sip.c
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r266579 | pabelanger | 2010-06-01 09:54:05 -0500 (Tue, 01 Jun 2010) | 18
lines
Missing fallback to audio fax feature when T.38 re-INVITE failed
When a T.38 re-INVITE failed with an 488 or 606 answer, we should
fallback to audio fax by send a re-re-INVITE without T.38. The
function is backported from 1.6 asterisk.
(closes issue https://issues.asterisk.org/view.php?id=16795)
Reported by: vrban
(closes issue https://issues.asterisk.org/view.php?id=16692)
Reported by: vrban
Patches:
t38_fallback_to_audio_v3.patch uploaded by vrban (license 756)
Tested by: lmadsen, vrban, haggard
https://reviewboard.asterisk.org/r/514/
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http://svn.digium.com/view/asterisk?view=rev&revision=266579
Issue History
Date Modified Username Field Change
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2010-06-01 09:54 svnbot Checkin
2010-06-01 09:54 svnbot Note Added: 0122687
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