[asterisk-bugs] [Asterisk 0016540]: Asterisk says "No compatible codecs, not accepting this offer!" on T.38 offer
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Jan 4 10:47:00 CST 2010
The following issue requires your FEEDBACK.
======================================================================
https://issues.asterisk.org/view.php?id=16540
======================================================================
Reported By: schogge
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 16540
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.4.28
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-01-04 10:39 CST
Last Modified: 2010-01-04 10:47 CST
======================================================================
Summary: Asterisk says "No compatible codecs, not accepting
this offer!" on T.38 offer
Description:
Hi community,
first of all: this is the first bug report in my life.
Since I updated asterisk from 1.4.25 to 1.4.28 I get the the following
message when I try to send or receive a fax with T.38 through asterisk:
"NOTICE[11689]: chan_sip.c:5621 process_sdp: No compatible codecs, not
accepting this offer!"
This is my setup for receiving faxes:
Provider with T.38 support ---SIP---> Asterisk with T.38 pass through
---SIP---> Callweaver with T.38 support
This is my setup for sending faxes:
Linksys SPA 2102 with T.38 support ---SIP---> Asterisk with T.38 pass
through ---SIP---> Provider with T.38 support
With asterisk version 1.4.25 I never saw this message.
Seems like this issue already existed in the past. See
https://issues.asterisk.org/view.php?id=12414
Please let me know if there is any further information needed.
Rgs,
schogge
======================================================================
----------------------------------------------------------------------
(0115930) lmadsen (administrator) - 2010-01-04 10:47
https://issues.asterisk.org/view.php?id=16540#c115930
----------------------------------------------------------------------
Per the bug guidelines, any SIP issues need to provide at least the
following documentation:
* SIP debug
* Console output
* SIP history
These can be enabled via the sip.conf file, and the 'core set verbose 10'
option on your console while you reproduce the issue.
Issue History
Date Modified Username Field Change
======================================================================
2010-01-04 10:47 lmadsen Note Added: 0115930
2010-01-04 10:47 lmadsen Status new => feedback
======================================================================
More information about the asterisk-bugs
mailing list