[asterisk-dev] [Code Review] SIP: port configuration
David Vossel
dvossel at digium.com
Mon Sep 21 15:07:07 CDT 2009
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/357/
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(Updated 2009-09-21 15:07:07.198210)
Review request for Asterisk Developers.
Changes
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Fixes sip_register() port logic to set the default port number correctly when no port is specified as well as when a port parsing error occurs.
Summary
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In chan_sip.c, build_peer() was built with the assumption that the default port for a peer would always be 5060. This is not the case when transport=tls. Now in build_peer() instead of setting the default port at the beginning of the function, the port is cleared until after option parsing is complete. This allows the correct default port to be set at the end of the function according to what transport type is specified. If a peer is registered, the peer's port will not be cleared or overridden during a reload. I also changed the "port" option parsing to check for errors by using sscanf() rather than atoi().
This addresses bug 15854.
https://issues.asterisk.org/view.php?id=15854
Diffs (updated)
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/trunk/channels/chan_sip.c 219749
Diff: https://reviewboard.asterisk.org/r/357/diff
Testing
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made a test call via tls, verified port 5061 is used by default rather than 5060
Thanks,
David
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