[asterisk-bugs] [Asterisk 0016061]: [patch] Peer mismatch in incomming call [PATCH]
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Feb 16 11:37:47 CST 2010
The following issue has been ASSIGNED.
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https://issues.asterisk.org/view.php?id=16061
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Reported By: dveiga
Assigned To:
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Project: Asterisk
Issue ID: 16061
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: assigned
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 223801
Request Review:
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Date Submitted: 2009-10-12 18:08 CDT
Last Modified: 2010-02-16 11:37 CST
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Summary: [patch] Peer mismatch in incomming call [PATCH]
Description:
I'm using the latest 1.4 svn version as a voip gateway connected to a
PBX. As such I have as many DAHDI channels as SIP peers and link them on a
one to one basis. When commertial gateways (say Grandstream 4024) register
to a PBX, they use different local SIP ports for each channel. In this way,
when the PBX places a SIP invite to offer a call using a certain port, ther
is no doubt which SIP peer the call is for.
Asterisk cannot use different ports, so upon receiving an invite, it
has to determine which peer it is for. The default behaviour is to
determine the peer based on the IP address (after trying to find a user,
that is not applicable here), but when there are multiple subscriptions to
the same IP this is not enough.
The current patch analyzes the sip headers to determine the peer name
and then finds the peer based on both: IP an NAME. For compatibility
reasons, if a peer matching both is not found, it falls back to the
previous method (finding one just using the IP address).
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Issue History
Date Modified Username Field Change
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2010-02-16 11:37 dvossel Assigned To dvossel =>
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