[asterisk-bugs] [Asterisk 0017072]: Asterisk 1.6.1.18 and 1.6.2.6 RTP BUG
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Mar 22 09:13:17 CDT 2010
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=17072
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Reported By: kvveltho
Assigned To:
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Project: Asterisk
Issue ID: 17072
Category: Core/General
Reproducibility: always
Severity: block
Priority: normal
Status: closed
Asterisk Version: 1.6.1.18
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: duplicate
Duplicate: 0
Fixed in Version:
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Date Submitted: 2010-03-22 08:11 CDT
Last Modified: 2010-03-22 09:13 CDT
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Summary: Asterisk 1.6.1.18 and 1.6.2.6 RTP BUG
Description:
When having more load (about 60 CC calls), the following message start the
flood the console:
[Mar 22 11:43:43] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:43] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
[Mar 22 11:43:44] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:44] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
[Mar 22 11:43:45] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:45] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
[Mar 22 11:43:45] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:45] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
[Mar 22 11:43:46] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:46] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
We have plenty of RTP ports both on asterisk and our router. See attached
rtp.conf
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Relationships ID Summary
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has duplicate 0017070 asterisk is not closing unused RTP ports
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Issue History
Date Modified Username Field Change
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2010-03-22 09:13 lmadsen Resolution open => duplicate
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