[Asterisk-code-review] pjsip channel: Use specified media port (testsuite[13])

[Corey Farrell]

Corey Farrell has posted comments on this change. ( https://gerrit.asterisk.org/8857 )

Change subject: pjsip_channel: Use specified media port
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Patch Set 1: Code-Review-1

(1 comment)

https://gerrit.asterisk.org/#/c/8857/1/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml
File tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml:

https://gerrit.asterisk.org/#/c/8857/1/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml@25
PS1, Line 25:       m=audio 9050 RTP/AVP 0
Shouldn't this be left alone, '-mp 9050' added to the scenario key-args instead?



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Gerrit-Project: testsuite
Gerrit-Branch: 13
Gerrit-MessageType: comment
Gerrit-Change-Id: Id7f82d8245401659d269e6ba084c2ad05aee068d
Gerrit-Change-Number: 8857
Gerrit-PatchSet: 1
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Corey Farrell <git at cfware.com>
Gerrit-Comment-Date: Tue, 24 Apr 2018 17:35:16 +0000
Gerrit-HasComments: Yes
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