[asterisk-commits] pjsip channel: Use specified media port (testsuite[master])

[SVN commits to the Asterisk project]

Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/8860 )

Change subject: pjsip_channel: Use specified media port
......................................................................

pjsip_channel: Use specified media port

This test took advantage of the fact that SIPp defaults the media port to 6000,
which it then checked against for the test. However, a recent patch made it so
tests using the SIPp 'media_port' option would get an available random port
instead.

This patch specifies the media_port for SIPp to use and makes sure it is the
same one checked by the test. A number was chosen which hopefully makes port
collisions highly unlikely.

Change-Id: Id7f82d8245401659d269e6ba084c2ad05aee068d
---
M tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf
M tests/channels/pjsip/dialplan_functions/pjsip_channel/test-config.yaml
2 files changed, 4 insertions(+), 3 deletions(-)

Approvals:
  Corey Farrell: Looks good to me, but someone else must approve
  Richard Mudgett: Looks good to me, approved
  Jenkins2: Approved for Submit



diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf b/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf
index 22ec556..8d6b12d 100644
--- a/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf
@@ -22,13 +22,13 @@
 
 ; Source will often be various things; just make sure we get something back
 same => n,GoSub(default,test_variable,1(rtp,src,audio,!=,""))
-same => n,GoSub(default,test_variable,1(rtp,dest,audio,=,"127.0.0.1:6000"))
+same => n,GoSub(default,test_variable,1(rtp,dest,audio,=,"127.0.0.1:9050"))
 same => n,GoSub(default,test_variable,1(rtp,hold,audio,=,"0"))
 same => n,GoSub(default,test_variable,1(rtp,secure,audio,=,"0"))
 same => n,GoSub(default,test_variable,1(rtp,direct,audio,=,"(null)"))
 
 ; Verify audio is set by default
-same => n,GoSub(default,test_variable,1(rtp,dest,,=,"127.0.0.1:6000"))
+same => n,GoSub(default,test_variable,1(rtp,dest,,=,"127.0.0.1:9050"))
 
 ; No video stream, these should be empty
 same => n,GoSub(default,test_variable,1(rtp,src,video,=,""))
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel/test-config.yaml b/tests/channels/pjsip/dialplan_functions/pjsip_channel/test-config.yaml
index 911edc6..8095c81 100644
--- a/tests/channels/pjsip/dialplan_functions/pjsip_channel/test-config.yaml
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel/test-config.yaml
@@ -20,7 +20,8 @@
     test-iterations:
         -
              scenarios:
-                - { 'key-args': { 'scenario': 'uac-no-hangup.xml', '-p': '5062', '-i': '127.0.0.1', '-s': 'alice', '-rsa': '127.0.0.1:5061', '-s': 'alice'} }
+                - { 'key-args': { 'scenario': 'uac-no-hangup.xml', '-p': '5062', '-i': '127.0.0.1',
+                '-s': 'alice', '-rsa': '127.0.0.1:5061', '-s': 'alice', '-mp': '9050'} }
 
 
 ami-config:

-- 
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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: Id7f82d8245401659d269e6ba084c2ad05aee068d
Gerrit-Change-Number: 8860
Gerrit-PatchSet: 3
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Corey Farrell <git at cfware.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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