[asterisk-bugs] [Asterisk 0010352]: When I allow the codec G729 first in sip.conf the other codecs are not offered in the INVITE
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Aug 1 07:46:42 CDT 2007
The following issue requires your FEEDBACK.
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http://bugs.digium.com/view.php?id=10352
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Reported By: Antxoneti
Assigned To:
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Project: Asterisk
Issue ID: 10352
Category: Codecs/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.4.9
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 08-01-2007 01:17 CDT
Last Modified: 08-01-2007 07:46 CDT
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Summary: When I allow the codec G729 first in sip.conf the
other codecs are not offered in the INVITE
Description:
I have made to test:
1- sip.conf
disallow = all
allow = g729
allow = alaw
allow = ulaw
And the trace of the INVITE is
m=audio 12480 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
No more codecs are offered than the G729
2- sip.conf
disallow = all
allow = alaw
allow = ulaw
allow = g729
And the trace INVITE is:
m=audio 19360 RTP/AVP 8 0 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
The codec G729 is not offered
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file - 08-01-07 07:46
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Please provide a full SIP debug when reporting SIP related bugs, and also a
description of the call flow.
Issue History
Date Modified Username Field Change
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08-01-07 07:46 file Note Added: 0068187
08-01-07 07:46 file Status new => feedback
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