[asterisk-bugs] [Asterisk 0017805]: SIP REFER auth fails, RTP timeout ignored, and other discrepancies between e/ingress calls
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Aug 6 12:53:37 CDT 2010
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=17805
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Reported By: kkm
Assigned To:
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Project: Asterisk
Issue ID: 17805
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 281083
Request Review:
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Date Submitted: 2010-08-06 01:22 CDT
Last Modified: 2010-08-06 12:53 CDT
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Summary: SIP REFER auth fails, RTP timeout ignored, and other
discrepancies between e/ingress calls
Description:
The following per-peer parameters in sip.conf are ignored for incoming
calls, but applied correctly to outgoing calls:
auth= :: if incoming call is transferred by Transfer() dialplan
application, user name and secret not used when constructing the digest in
reply to the 407 response.
rtptimeout= :: overrides global value of the same for outgoing calls but
not for incoming calls.
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Issue History
Date Modified Username Field Change
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2010-08-06 12:53 lmadsen Description Updated
2010-08-06 12:53 lmadsen Additional Information Updated
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