[asterisk-dev] [Code Review]: Testsuite: Disallow MoH upon hold option
Matt Jordan
reviewboard at asterisk.org
Thu Feb 21 20:50:56 CST 2013
> On Feb. 21, 2013, 4:59 p.m., opticron wrote:
> > /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_media_restrict.xml, lines 8-14
> > <https://reviewboard.asterisk.org/r/2337/diff/1/?file=33505#file33505line8>
> >
> > Picking out the Call-ID is unnecessary in this case. [call_id] or [last_Call-ID] will work for this purpose.
I'm not sure they will work in this scenario. If I recall correctly, [call_id] generates a different Call-ID then the received INVITE request, which makes the subsequent INVITE request that the scenario sends a different dialog. You can try [last_Call-ID], but I vaguely remember that SIPp doesn't handle re-INVITEs properly in this regard.
It's possible I'm wrong so it's worth trying, but when I first wrote the SIPp hold tests I spent quite awhile hacking around on the Call-ID trying to get the re-INVITE to work.
- Matt
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On Feb. 18, 2013, 2:30 p.m., Kevin Harwell wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2337/
> -----------------------------------------------------------
>
> (Updated Feb. 18, 2013, 2:30 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Summary
> -------
>
> Tests to make sure the music on hold event is not triggered if the "discard_remote_hold_retrieval" option is set to "yes". In the test multiple scenarios are ran where one SIP phone puts another SIP phone on hold by sending a re-INVITE with a modified SDP containing either a restricted audio direction, an IP address of 0.0.0.0, or a combination thereof. This is tested both for a local RTP bridge, and a non-bridged scenario.
>
>
> This addresses bug ABE-2899.
> https://issues.asterisk.org/jira/browse/ABE-2899
>
>
> Diffs
> -----
>
> /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/configs/ast1/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/configs/ast1/sip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_A.xml PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_IP_media_restrict.xml PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_IP_restrict.xml PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_media_restrict.xml PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/test-config.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/tests.yaml 3642
>
> Diff: https://reviewboard.asterisk.org/r/2337/diff
>
>
> Testing
> -------
>
> Ran the test and made sure all scenarios passed. Also set the discard_remote_hold_retrieval to "no" and made sure the test failed.
>
>
> Thanks,
>
> Kevin
>
>
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