Author: mjordan Date: Wed Dec 24 06:55:11 2014 New Revision: 6135 URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=6135 Log: Add a test for user_eq_phone setting in PJSIP
This patch adds a test for the user_eq_phone endpoint setting in PJSIP. The test verifies that when the user_eq_phone setting is enabled on a PJSIP endpoint, a request sent from Asterisk to that endpoint that contains a telephone number in the request URI has a 'user=phone' specified appended to it. The test originates a Local channel that causes an outbound dial to number 12568675309 at endpoint 'jenny'. The SIPp scenario verifies that a 'user=phone' tag is found in the INVITE request received from Asterisk. Review: https://reviewboard.asterisk.org/r/4294/ Added: asterisk/trunk/tests/channels/pjsip/user_eq_phone/ asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/ asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/extensions.conf (with props) asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/pjsip.conf (with props) asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/ asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/uas.xml (with props) asterisk/trunk/tests/channels/pjsip/user_eq_phone/test-config.yaml (with props) Modified: asterisk/trunk/tests/channels/pjsip/tests.yaml Modified: asterisk/trunk/tests/channels/pjsip/tests.yaml URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/tests.yaml?view=diff&rev=6135&r1=6134&r2=6135 ============================================================================== --- asterisk/trunk/tests/channels/pjsip/tests.yaml (original) +++ asterisk/trunk/tests/channels/pjsip/tests.yaml Wed Dec 24 06:55:11 2014 @@ -30,4 +30,5 @@ - dir: 'optimistic_srtp' - test: 'in_dialog_invite_replaces' - test: 'dtmf_incompatible' + - test: 'user_eq_phone' - test: 'keep_alive' Added: asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/extensions.conf URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/extensions.conf?view=auto&rev=6135 ============================================================================== --- asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/extensions.conf (added) +++ asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/extensions.conf Wed Dec 24 06:55:11 2014 @@ -1,0 +1,5 @@ +[default] + +exten => s,1,NoOp() + same => n,Dial(PJSIP/+12568675309@jenny) + same => n,Hangup() Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/extensions.conf ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/extensions.conf ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/extensions.conf ------------------------------------------------------------------------------ svn:mime-type = text/plain Added: asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/pjsip.conf URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/pjsip.conf?view=auto&rev=6135 ============================================================================== --- asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/pjsip.conf (added) +++ asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/pjsip.conf Wed Dec 24 06:55:11 2014 @@ -1,0 +1,16 @@ + +[transport-udp] +type=transport +protocol=udp +bind=0.0.0.0:5060 + +[jenny] +type=endpoint +context=default +allow=!all,ulaw,alaw,g722 +user_eq_phone=True +aors=jenny + +[jenny] +type=aor +contact=sip:127.0.0.1:5061 Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/pjsip.conf ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/pjsip.conf ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/configs/ast1/pjsip.conf ------------------------------------------------------------------------------ svn:mime-type = text/plain Added: asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/uas.xml URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/uas.xml?view=auto&rev=6135 ============================================================================== --- asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/uas.xml (added) +++ asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/uas.xml Wed Dec 24 06:55:11 2014 @@ -1,0 +1,70 @@ +<?xml version="1.0" encoding="ISO-8859-1" ?> +<!DOCTYPE scenario SYSTEM "sipp.dtd"> + +<scenario name="Receive INVITE with video"> + <Global variables="remote_tag" /> + <recv request="INVITE" crlf="true"> + <action> + <!-- Save the from tag. We'll need it when we send our BYE --> + <ereg regexp=".*(;tag=.*)" + header="From:" + search_in="hdr" + check_it="true" + assign_to="remote_tag"/> + <ereg regexp=".*;user=phone.*" + search_in="msg" + check_it="true" + assign_to="1"/> + </action> + </recv> + <Reference variables="1" /> + + <send retrans="500"> + <![CDATA[ + + SIP/2.0 200 OK + [last_Via:] + [last_From:] + [last_To:];tag=[call_number] + [last_Call-ID:] + [last_CSeq:] + Contact: <sip:[local_ip]:[local_port];transport=[transport]> + Content-Type: application/sdp + Content-Length: [len] + + v=0 + o=- 1324901698 1324901698 IN IP4 [local_ip] + s=- + c=IN IP4 [local_ip] + t=0 0 + m=audio 2226 RTP/AVP 0 101 + a=sendrecv + a=rtpmap:0 PCMU/8000 + a=rtpmap:101 telephone-event/8000 + + ]]> + </send> + + <recv request="ACK" rtd="true" crlf="true"> + </recv> + + <send retrans="500"> + <![CDATA[ + + BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 + Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] + From: ua1 <sip:ua1@[local_ip]:[local_port]>;tag=[call_number] + To: [$remote_tag] + [last_Call-ID:] + CSeq: [cseq] BYE + Contact: <sip:[local_ip]:[local_port];transport=[transport]> + Max-Forwards: 70 + Content-Length: 0 + + ]]> + </send> + + <recv response="200"> + </recv> + +</scenario> Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/uas.xml ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/uas.xml ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/sipp/uas.xml ------------------------------------------------------------------------------ svn:mime-type = text/plain Added: asterisk/trunk/tests/channels/pjsip/user_eq_phone/test-config.yaml URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/user_eq_phone/test-config.yaml?view=auto&rev=6135 ============================================================================== --- asterisk/trunk/tests/channels/pjsip/user_eq_phone/test-config.yaml (added) +++ asterisk/trunk/tests/channels/pjsip/user_eq_phone/test-config.yaml Wed Dec 24 06:55:11 2014 @@ -1,0 +1,41 @@ +testinfo: + summary: 'Verify the user_eq_phone setting' + description: | + This test verifies that when the user_eq_phone setting is enabled on a + PJSIP endpoint, a request sent from Asterisk to that endpoint that + contains a telephone number in the request URI has a 'user=phone' + specified appended to it. The test originates a Local channel that + causes an outbound dial to number 12568675309 at endpoint 'jenny'. + The SIPp scenario verifies that a 'user=phone' tag is found in the + INVITE request received from Asterisk. + +test-modules: + test-object: + config-section: test-object-config + typename: 'sipp.SIPpTestCase' + modules: + - + config-section: originator + typename: 'pluggable_modules.Originator' + +test-object-config: + test-iterations: + - + scenarios: + - { 'key-args': {'scenario': 'uas.xml', '-i': '127.0.0.1', '-p': '5061'} } + +originator: + trigger: 'ami_connect' + ignore-originate-failure: 'no' + id: '0' + channel: 'Local/s@default' + application: 'Echo' + async: 'True' + +properties: + minversion: '13.2.0' + dependencies: + - app : 'sipp' + - asterisk : 'res_pjsip' + tags: + - pjsip Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/test-config.yaml ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/test-config.yaml ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/pjsip/user_eq_phone/test-config.yaml ------------------------------------------------------------------------------ svn:mime-type = text/plain -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- svn-commits mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/svn-commits
