<p>Kevin Harwell has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/8858">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">pjsip_channel: Use specified media port<br><br>This test took advantage of the fact that SIPp defaults the media port to 6000,<br>which it then checked against for the test. However, a recent patch made it so<br>tests using the SIPp 'media_port' option would get an available random port<br>instead.<br><br>This patch no longer uses the 'media_port' option, but instead uses a hard<br>coded value for the media port in the SDP. A number was chosen which hopefully<br>makes port collisions highly unlikely.<br><br>Change-Id: Id7f82d8245401659d269e6ba084c2ad05aee068d<br>---<br>M tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf<br>M tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml<br>2 files changed, 3 insertions(+), 3 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/58/8858/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">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<br>index 22ec556..8d6b12d 100644<br>--- a/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf<br>+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf<br>@@ -22,13 +22,13 @@<br> <br> ; Source will often be various things; just make sure we get something back<br> same => n,GoSub(default,test_variable,1(rtp,src,audio,!=,""))<br>-same => n,GoSub(default,test_variable,1(rtp,dest,audio,=,"127.0.0.1:6000"))<br>+same => n,GoSub(default,test_variable,1(rtp,dest,audio,=,"127.0.0.1:9050"))<br> same => n,GoSub(default,test_variable,1(rtp,hold,audio,=,"0"))<br> same => n,GoSub(default,test_variable,1(rtp,secure,audio,=,"0"))<br> same => n,GoSub(default,test_variable,1(rtp,direct,audio,=,"(null)"))<br> <br> ; Verify audio is set by default<br>-same => n,GoSub(default,test_variable,1(rtp,dest,,=,"127.0.0.1:6000"))<br>+same => n,GoSub(default,test_variable,1(rtp,dest,,=,"127.0.0.1:9050"))<br> <br> ; No video stream, these should be empty<br> same => n,GoSub(default,test_variable,1(rtp,src,video,=,""))<br>diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml b/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml<br>index ef10798..b3c3c89 100644<br>--- a/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml<br>+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml<br>@@ -22,7 +22,7 @@<br>       s=-<br>       c=IN IP[media_ip_type] [media_ip]<br>       t=0 0<br>-      m=audio [media_port] RTP/AVP 0<br>+      m=audio 9050 RTP/AVP 0<br>       a=rtpmap:0 PCMU/8000<br> <br>     ]]><br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/8858">change 8858</a>. To unsubscribe, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/8858"/><meta itemprop="name" content="View Change"/></div></div>

<div style="display:none"> Gerrit-Project: testsuite </div>
<div style="display:none"> Gerrit-Branch: 14 </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>
<div style="display:none"> Gerrit-Change-Id: Id7f82d8245401659d269e6ba084c2ad05aee068d </div>
<div style="display:none"> Gerrit-Change-Number: 8858 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Kevin Harwell <kharwell@digium.com> </div>