[asterisk-dev] [Code Review] Check for ANY to place calls on hold for SIP

[mjordan]

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Review request for Asterisk Developers and Tzafrir Cohen.


Summary
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As reported, Asterisk (1.6.2 and prior) would previously place calls on hold if an INVITE was received with c=0.0.0.0.  This fails in 1.8 and later, as we only check to see if the various addresses are null.

This patch resolves this by checking if the addresses received are null or any.  This is done using the netsock2 library of functions, so it should be compatible with both IPv4 and IPv6.


This addresses bug ASTERISK-18086.
    https://issues.asterisk.org/jira/browse/ASTERISK-18086


Diffs
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  /branches/1.8/channels/chan_sip.c 340135 

Diff: https://reviewboard.asterisk.org/r/1504/diff


Testing
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Tested using a cobbled together SIPp scenario, which appeared to behave as expected.  Tested also under the Asterisk TestSuite, which continued to work as expected.


Thanks,

mjordan

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