[asterisk-dev] [Code Review] SIP Blind transfer tests

[Matt Jordan]

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https://reviewboard.asterisk.org/r/1686/
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Review request for Asterisk Developers and Mark Michelson.


Summary
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This patch adds four SIP blind transfer tests to the testsuite.  For Phone A, Phone B, and Phone C, where Phone A initially calls Phone B, they test:
1. Phone A initiating a blind transfer of Phone B to Phone C with no re-INVITE prior to the REFER message
2. Phone A initiating a blind transfer of Phone B to Phone C with a re-INVITE
3. Phone B initiating a blind transfer of Phone A to Phone C with no re-INVITE prior to the REFER message
4. Phone B initiating a blind transfer of Phone A to Phone C with a re-INVITE

Note that adding the 'h' extension currently reproduces the bug (ASTERISK-19173) fixed by Mark on patch https://reviewboard.asterisk.org/r/1685/, hence its inclusion in the test.


This addresses bug ASTERISK-19173.
    https://issues.asterisk.org/jira/browse/ASTERISK-19173


Diffs
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  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/configs/ast1/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/configs/ast1/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/test-config.yaml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/configs/ast1/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/configs/ast1/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/run-test PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/test-config.yaml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_refer_only/configs/ast1/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_refer_only/configs/ast1/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_refer_only/run-test PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_refer_only/test-config.yaml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/configs/ast1/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/configs/ast1/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/run-test PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/test-config.yaml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/tests.yaml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/tests.yaml 3000 

Diff: https://reviewboard.asterisk.org/r/1686/diff


Testing
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Need to test with Mark's patch - without the patch, Scenarios 3 and 4 will fail, while Scenario 1 results in an orphaned bridge.  It is expected that the patch resolves all three of those issues.


Thanks,

Matt

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