[Asterisk-code-review] res pjsip: Call Transfer with Empty Extension (testsuite[master])
Anonymous Coward
asteriskteam at digium.com
Mon Mar 20 13:47:52 CDT 2017
Anonymous Coward #1000019 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5152 )
Change subject: res_pjsip: Call Transfer with Empty Extension
......................................................................
res_pjsip: Call Transfer with Empty Extension
When performing a call transfer to a sip uri w/o a username part
make sure that the 's' extension of the dialplan is invoked
ASTERISK-26869
Change-Id: Iba651ccc278ebd18ba854b4d85b986ccb7c0e6ba
---
A tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/configs/ast1/extensions.conf
A tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/configs/ast1/pjsip.conf
A tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/sipp/transferer.xml
A tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/test-config.yaml
M tests/channels/pjsip/transfers/blind_transfer/tests.yaml
5 files changed, 180 insertions(+), 1 deletion(-)
Approvals:
Kevin Harwell: Looks good to me, approved
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/configs/ast1/extensions.conf b/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/configs/ast1/extensions.conf
new file mode 100644
index 0000000..a6d7535
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/configs/ast1/extensions.conf
@@ -0,0 +1,12 @@
+[default]
+
+exten => echo,1,Answer()
+same => n,Echo()
+same => n,Hangup()
+
+exten => call-sipp,1,Dial(PJSIP/sipp)
+same => n,Hangup()
+
+exten => s,1,Answer()
+same => n,UserEvent(Transferred)
+same => n,Hangup()
diff --git a/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/configs/ast1/pjsip.conf b/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..7132032
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/configs/ast1/pjsip.conf
@@ -0,0 +1,14 @@
+[main-transport]
+type = transport
+protocol = udp
+bind = 127.0.0.1
+
+[sipp]
+type = endpoint
+context = default
+allow = ulaw
+aors = sipp
+
+[sipp]
+type = aor
+contact = sip:sipp at 127.0.0.1:5061
diff --git a/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/sipp/transferer.xml b/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/sipp/transferer.xml
new file mode 100644
index 0000000..1737d1e
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/sipp/transferer.xml
@@ -0,0 +1,90 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="transferer">
+ <Global variables="remote_tag" />
+ <recv request="INVITE" crlf="true">
+ <!-- Save the from tag. We'll need it when we send our REFER -->
+ <action>
+ <ereg regexp="(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag00[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=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <pause milliseconds="2000" />
+
+ <!-- Blind transfer this sucker to, well, nowhere which should be s -->
+ <send retrans="500">
+ <![CDATA[
+
+ REFER sip:sipp@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[$remote_tag]
+ Call-ID: [call_id]
+ CSeq: [cseq] REFER
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Refer-To: sip:[remote_ip]:[remote_port];user=phone
+ Referred-By: sip:sipp@[local_ip]:[local_port]
+ Content-Length: 0
+
+ ]]>
+
+ </send>
+
+ <recv response="202" rtd="true">
+ </recv>
+
+ <recv request="NOTIFY" crlf="true">
+ </recv>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/test-config.yaml b/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/test-config.yaml
new file mode 100644
index 0000000..61bc622
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/no_target_in_refer/test-config.yaml
@@ -0,0 +1,63 @@
+testinfo:
+ summary: "Ensure that when a REFER without a user is received we transfer to s extension"
+ description: |
+ 'Asterisk originates a call to a Local channel that runs Echo. The other half of
+ the local channel is placed into the dialplan and calls a SIPp scenario. The SIPp
+ scenario answers the call and then performs a blind transfer without specifying
+ an extension in the REFER. The blind transfer should go to the s extension and be
+ considered successful. An event which is emitted by the dialplan is used to determine
+ if the s extension was executed and to end the test.
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: sipp.SIPpTestCase
+ modules:
+ -
+ config-section: originator-config
+ typename: pluggable_modules.Originator
+ -
+ config-section: pluggable-config
+ typename: 'pluggable_modules.EventActionModule'
+
+sipp-config:
+ stop-after-scenarios: false
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'transferer.xml', '-i': '127.0.0.1', '-p': '5061' }}
+
+originator-config:
+ channel: 'Local/echo at default'
+ context: 'default'
+ exten: 'call-sipp'
+ priority: '1'
+ trigger: 'scenario_start'
+ scenario-name: 'transferer.xml'
+
+pluggable-config:
+ -
+ ami-events:
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'Transferred'
+ count: 1
+ stop_test:
+
+properties:
+ minversion: '13.15.0'
+ dependencies:
+ - sipp:
+ version: 'v3.0'
+ - asterisk: 'res_pjsip'
+ - asterisk: 'res_pjsip_session'
+ - asterisk: 'res_pjsip_refer'
+ - asterisk: 'chan_pjsip'
+ - asterisk: 'app_dial'
+ - asterisk: 'app_echo'
+ tags:
+ - pjsip
+ testconditions:
+ -
+ name: 'channels'
diff --git a/tests/channels/pjsip/transfers/blind_transfer/tests.yaml b/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
index a529add..6688906 100644
--- a/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
+++ b/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
@@ -9,4 +9,4 @@
- test: 'caller_with_hold'
- test: 'caller_with_hold_drop_options'
- test: 'goto_on_blindxfr'
-
+ - test: 'no_target_in_refer'
--
To view, visit https://gerrit.asterisk.org/5152
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: Iba651ccc278ebd18ba854b4d85b986ccb7c0e6ba
Gerrit-PatchSet: 4
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
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