[asterisk-commits] testsuite: Add a test for PJSIP DTMF MODE (testsuite[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Aug 1 14:51:31 CDT 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5910 )
Change subject: testsuite: Add a test for PJSIP_DTMF_MODE
......................................................................
testsuite: Add a test for PJSIP_DTMF_MODE
This adds a test for the new PJSIP_DTMF_MODE
dialplan function which allows querying and changing
the DTMF mode of a PJSIP channel in the dialplan.
ASTERISK-27085
Change-Id: I8ae7d73aedc42533512adf7b96ed0eab8e09ad8c
---
A tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/configs/ast1/extensions.conf
A tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/configs/ast1/pjsip.conf
A tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/sipp/A_PARTY.xml
A tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/test-config.yaml
M tests/channels/pjsip/dialplan_functions/tests.yaml
5 files changed, 232 insertions(+), 0 deletions(-)
Approvals:
Joshua Colp: Looks good to me, approved; Verified
Jenkins2: Approved for Submit
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/configs/ast1/extensions.conf b/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/configs/ast1/extensions.conf
new file mode 100644
index 0000000..24097ee
--- /dev/null
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/configs/ast1/extensions.conf
@@ -0,0 +1,14 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+exten => _X.,1,Set(PJSIP_DTMF_MODE()=info)
+exten => _X.,n,Answer()
+exten => _X.,n,SendDTMF(4)
+exten => _X.,n,Wait(360)
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/configs/ast1/pjsip.conf b/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..995fccb
--- /dev/null
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/configs/ast1/pjsip.conf
@@ -0,0 +1,55 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 6060
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+debug = yes
+
+[transport-udp6]
+type = transport
+protocol = udp
+bind = [::]:5060
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:5060
+
+[PEER_A]
+type = aor
+contact = sip:127.0.0.1:6060
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+disallow = all
+allow = alaw
+allow = ulaw
+allow = g729
+allow = h263p
+allow = h264
+direct_media = no
+send_rpid = yes
+sdp_session = session
+aors = PEER_A
+t38_udptl = yes
+t38_udptl_ec = redundancy
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/sipp/A_PARTY.xml b/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/sipp/A_PARTY.xml
new file mode 100644
index 0000000..c8c445d
--- /dev/null
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/sipp/A_PARTY.xml
@@ -0,0 +1,133 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp 'uac' scenario with pcap (rtp) play -->
+<!-- -->
+<scenario name="DTMF_INFO_FORCE">
+ <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
+ <!-- generated by sipp. To do so, use [call_id] keyword. -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test at voxbone.com>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 9000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-15
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <!-- By adding rrs="true" (Record Route Sets), the route sets -->
+ <!-- are saved and used for following messages sent. Useful to test -->
+ <!-- against stateful SIP proxies/B2BUAs. -->
+ <recv response="200" rtd="true" crlf="true">
+ </recv>
+
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+<!-- Receive the DIGIT 4-->
+<recv request="INFO">
+ <action>
+ <ereg regexp="(Signal=4)" search_in="body" check_it="true" assign_to = "1" />
+ <log message="---DTMF--- [$1]"/>
+ </action>
+</recv>
+
+
+<send>
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_From:]
+[last_To:]
+[last_Call-ID:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/test-config.yaml b/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/test-config.yaml
new file mode 100644
index 0000000..f6b969f
--- /dev/null
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_dtmfmode/test-config.yaml
@@ -0,0 +1,29 @@
+testinfo:
+ summary: 'Receive a SIP call and confirm that changing the DTMF mode works'
+ description: |
+ 'Using SIPp place a call into Asterisk that negotiates DTMF using RFC2833. In the dialplan change
+ the DTMF mode using the PJSIP_DTMF_MODE dialplan function. Once changed send a DTMF using SendDTMF
+ and confirm that INFO DTMF was sent instead of the negotiated RFC2833.'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'A_PARTY.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '1000'} }
+
+properties:
+ minversion: '13.18.0'
+ dependencies:
+ - python: 'twisted'
+ - python: 'starpy'
+ - sipp:
+ version: 'v3.3'
+ - asterisk: 'res_pjsip'
+ tags:
+ - pjsip
+
diff --git a/tests/channels/pjsip/dialplan_functions/tests.yaml b/tests/channels/pjsip/dialplan_functions/tests.yaml
index 1aed251..f653aa8 100644
--- a/tests/channels/pjsip/dialplan_functions/tests.yaml
+++ b/tests/channels/pjsip/dialplan_functions/tests.yaml
@@ -7,3 +7,4 @@
- test: 'pjsip_session_refresh'
- test: 'chan_is_avail'
- test: 'pjsip_header'
+ - test: 'pjsip_dtmfmode'
--
To view, visit https://gerrit.asterisk.org/5910
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: I8ae7d73aedc42533512adf7b96ed0eab8e09ad8c
Gerrit-Change-Number: 5910
Gerrit-PatchSet: 9
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-commits/attachments/20170801/88422ffe/attachment-0001.html>
More information about the asterisk-commits
mailing list