[Asterisk-code-review] pjsip/transfer: Test "From" anonymization (testsuite[master])
Anonymous Coward
asteriskteam at digium.com
Thu Mar 10 05:48:22 CST 2016
Anonymous Coward #1000019 has submitted this change and it was merged.
Change subject: pjsip/transfer: Test "From" anonymization
......................................................................
pjsip/transfer: Test "From" anonymization
Because of the multiple call legs, a transfer test is ideal for testing the
anonymization of the From header when callerid presentation is prohibited.
Change-Id: If9ebc98bd60d9ae7d5c90ddc57ae1a3f5b4f8f90
---
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml
M tests/channels/pjsip/transfers/attended_transfer/nominal/tests.yaml
8 files changed, 813 insertions(+), 0 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, but someone else must approve
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, approved
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf
new file mode 100644
index 0000000..a299118
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf
@@ -0,0 +1,9 @@
+
+[default]
+exten => call_c,1,NoOp()
+ same => n,Dial(PJSIP/charlie)
+ same => n,Hangup()
+
+exten => alice,1,NoOp()
+ same => n,Dial(PJSIP/bob)
+ same => n,Hangup()
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..d7e4874
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf
@@ -0,0 +1,38 @@
+[local]
+type=transport
+protocol=udp
+bind=127.0.0.1:5060
+
+[endpoint](!)
+type=endpoint
+context=default
+disallow=all
+allow=ulaw
+direct_media=no
+send_pai=yes
+send_rpid=yes
+trust_id_outbound=yes
+trust_id_inbound=yes
+callerid_privacy=prohib
+
+[alice](endpoint)
+callerid=Alice <alice>
+
+[bob](endpoint)
+aors=bob
+callerid=Bob <bob>
+
+[bob]
+type=aor
+contact=sip:bob at 127.0.0.1:5066
+
+[charlie](endpoint)
+aors=charlie
+callerid=Charlie <charlie>
+
+[charlie]
+type=aor
+contact=sip:charlie at 127.0.0.1:5067
+
+[david](endpoint)
+callerid=David <david>
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
new file mode 100644
index 0000000..f4166e1
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
@@ -0,0 +1,137 @@
+<?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 -->
+<!-- -->
+
+<scenario name="Referee Leg">
+
+ <recvCmd>
+ <action>
+ <ereg regexp="REMOTE(.*)"
+ search_in="hdr"
+ header="Call-ID:"
+ check_it="true"
+ assign_to="1,original_callid" />
+ </action>
+ </recvCmd>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:call_c@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:transfer@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 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 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+ <recv response="101" optional="true" />
+ <recv response="180" optional="true" />
+ <recv response="200" rtd="true" crlf="true">
+ <action>
+ <ereg regexp="tag=([[:alnum:].\-]*)"
+ search_in="hdr"
+ header="To:"
+ check_it="true"
+ assign_to="2,to_tag" />
+ <ereg regexp="tag=([[:alnum:].\-]*)"
+ search_in="hdr"
+ header="From:"
+ check_it="true"
+ assign_to="3,from_tag" />
+ </action>
+ </recv>
+ <Reference variables="1,2,3" />
+
+ <send>
+ <![CDATA[
+
+ ACK sip:call_c@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+ [last_From:]
+ [last_To]
+ Call-ID: [call_id]
+ CSeq: [cseq] ACK
+ Contact: sip:bob@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="1000" />
+ <sendCmd>
+ <![CDATA[
+ Call-ID: [$original_callid]
+ Remote-To-Tag: [$to_tag]
+ Remote-From-Tag: [$from_tag]
+ Remote-URI: sip:call_c@[remote_ip]:[remote_port]
+ ]]>
+ </sendCmd>
+
+ <recv request="BYE">
+ <action>
+ <ereg regexp="<sip:transfer at 127.0.0.1>"
+ header="From"
+ search_in="hdr"
+ check_it="true"
+ assign_to="from"/>
+ </action>
+ </recv>
+ <Reference variables="from" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Content-Length:0
+
+ ]]>
+ </send>
+
+ <!-- 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/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml
new file mode 100644
index 0000000..950cc54
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml
@@ -0,0 +1,252 @@
+<?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 -->
+<!-- -->
+
+<scenario name="Referer Leg">
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
+ header="From"
+ search_in="hdr"
+ check_it="true"
+ assign_to="from"/>
+ </action>
+ </recv>
+
+ <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=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">
+ <action>
+ <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
+ header="From"
+ search_in="hdr"
+ check_it="true"
+ assign_to="from"/>
+ <ereg regexp=" (.+)"
+ search_in="hdr"
+ header="From:"
+ check_it="true"
+ assign_to="1,outbound_to_header" />
+ <ereg regexp=" (.+)"
+ search_in="hdr"
+ header="To:"
+ check_it="true"
+ assign_to="1,outbound_from_header" />
+ </action>
+ </recv>
+
+ <!-- Put this leg on hold -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];rport;received=127.0.0.1;branch=[branch]
+ From: [$outbound_from_header]
+ To: [$outbound_to_header]
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Max-Forwards: 70
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+ <recv response="101" optional="true" />
+ <recv response="180" optional="true" />
+ <recv response="200" rtd="true" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[local_ip]:[local_port] SIP/2.0
+ [last_Via]
+ [last_From]
+ [last_To]
+ Call-ID: [call_id]
+ CSeq: [cseq] ACK
+ Contact: sip:bob@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <sendCmd>
+ <![CDATA[
+ Call-ID: REMOTE[call_id]
+ Start the Echo Leg
+ ]]>
+ </sendCmd>
+
+ <recvCmd>
+ <action>
+ <ereg regexp=" (.+)"
+ search_in="hdr"
+ header="Remote-URI:"
+ check_it="true"
+ assign_to="1,remote_contact" />
+ <ereg regexp=" (.+)"
+ search_in="hdr"
+ header="Remote-To-Tag:"
+ check_it="true"
+ assign_to="2,remote_to_tag" />
+ <ereg regexp=" (.+)"
+ search_in="hdr"
+ header="Remote-From-Tag:"
+ check_it="true"
+ assign_to="3,remote_from_tag" />
+ </action>
+ </recvCmd>
+ <Reference variables="1,2,3" />
+
+ <send>
+ <![CDATA[
+
+ REFER sip:call_c@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ [last_From:]
+ [last_To]
+ [last_Call-ID:]
+ CSeq: [cseq] REFER
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Max-Forwards: 70
+ Refer-to: <[$remote_contact]?Replaces=REMOTE[call_id]%3Bto-tag%3D[$remote_to_tag]%3Bfrom-tag%3D[$remote_from_tag]>
+ Referred-By: sip:bob@[local_ip]
+ Content-Length: 0
+
+ ]]>
+ </send>
+ <recv response="202" rtd="true" crlf="true" />
+
+ <!-- In a nominal attended transfer Asterisk should always
+ be sending two notifies (SIP frags of 100 and 200) -->
+ <recv request="NOTIFY" >
+ <action>
+ <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
+ header="From"
+ search_in="hdr"
+ check_it="true"
+ assign_to="from"/>
+ </action>
+ </recv>
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Content-Length:0
+
+ ]]>
+ </send>
+
+ <recv request="NOTIFY">
+ <action>
+ <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
+ header="From"
+ search_in="hdr"
+ check_it="true"
+ assign_to="from"/>
+ </action>
+ </recv>
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Content-Length:0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:call_c@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:transfer@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: [cseq] BYE
+ Contact: sip:bob@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200"/>
+
+ <!-- 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"/>
+
+ <Reference variables="from" />
+
+</scenario>
+
+
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml
new file mode 100644
index 0000000..321e53f
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml
@@ -0,0 +1,141 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:alice@[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[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="181"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="INVITE">
+ <action>
+ <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
+ header="From"
+ search_in="hdr"
+ check_it="true"
+ assign_to="from"/>
+ <ereg regexp="\"Charlie\" <sip:charlie at 127.0.0.1>"
+ header="P-Asserted-Identity"
+ search_in="hdr"
+ check_it="true"
+ assign_to="asserted_identity"/>
+ <ereg regexp="\"Charlie\" <sip:charlie at 127.0.0.1>"
+ header="Remote-Party-ID"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_party_id"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [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">
+ </recv>
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <timewait milliseconds="4000"/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+ <Reference variables="asserted_identity" />
+ <Reference variables="remote_party_id" />
+ <Reference variables="from" />
+
+</scenario>
+
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml
new file mode 100644
index 0000000..69c014d
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml
@@ -0,0 +1,166 @@
+<?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 default 'uas' scenario. -->
+<!-- -->
+
+<scenario name="Basic UAS responder">
+ <!-- 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 request="INVITE" crlf="true">
+ </recv>
+
+ <!-- The '[last_*]' keyword is replaced automatically by the -->
+ <!-- specified header if it was present in the last message received -->
+ <!-- (except if it was a retransmission). If the header was not -->
+ <!-- present or if no message has been received, the '[last_*]' -->
+ <!-- keyword is discarded, and all bytes until the end of the line -->
+ <!-- are also discarded. -->
+ <!-- -->
+ <!-- If the specified header was present several times in the -->
+ <!-- message, all occurences are concatenated (CRLF seperated) -->
+ <!-- to be used in place of the '[last_*]' keyword. -->
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[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=[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"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv request="INVITE">
+ <action>
+ <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
+ header="From"
+ search_in="hdr"
+ check_it="true"
+ assign_to="from"/>
+ <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
+ header="P-Asserted-Identity"
+ search_in="hdr"
+ check_it="true"
+ assign_to="asserted_identity"/>
+ <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
+ header="Remote-Party-ID"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_party_id"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [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">
+ </recv>
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- Keep the call open for a while in case the 200 is lost to be -->
+ <!-- able to retransmit it if we receive the BYE again. -->
+ <pause milliseconds="4000"/>
+
+
+ <!-- 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"/>
+
+ <Reference variables="asserted_identity" />
+ <Reference variables="remote_party_id" />
+ <Reference variables="from" />
+
+</scenario>
+
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml
new file mode 100644
index 0000000..642336b
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml
@@ -0,0 +1,69 @@
+testinfo:
+ summary: Test anonymized From headers while performing a callee-initiated attended transfer.
+ description: |
+ "Start four SIPp scenarios that do the following:
+ SIPp #1 (uac-no-hangup.xml) calls through Asterisk to SIPp #2 (referer_uas.xml)
+ SIPp #2 kicks off SIPp #3 (referee.xml) which calls SIPp #4 (uas.xml).
+ SIPp #3 passes call information back to SIPp #2.
+ SIPp #2 initiates an attended transfer via REFER with Replaces information from SIPp #3.
+ SIPp #1 and SIPp #4 are bridged.
+ SIPp #1 and SIPp #4 receive connected line updates and the values are checked.
+ SIPp #2 and SIPp #3 are hung up.
+ SIPp #1 and SIPp #4 are hung up."
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: sipp.SIPpTestCase
+ modules:
+ -
+ config-section: ami-config
+ typename: 'pluggable_modules.EventActionModule'
+
+test-object-config:
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'coordinated-sender': {'key-args': {'scenario':'referer_uas.xml', '-p':'5066', '-sleep': '2'} },
+ 'coordinated-receiver': { 'key-args': {'scenario':'referee.xml', '-p':'5065'} } }
+ - { 'key-args': {'scenario':'uas.xml', '-p':'5067', '-sleep': '2'} }
+ - { 'key-args': {'scenario':'uac-no-hangup.xml', '-p':'5068', '-s':'alice', '-sleep': '2'} }
+
+ami-config:
+ -
+ ami-events:
+ type: 'headermatch'
+ conditions:
+ match:
+ Event: 'AttendedTransfer'
+ Result: 'Success'
+ count: 1
+ # Ensure COLP updates occur for alice and charlie before hanging up.
+ -
+ ami-events:
+ conditions:
+ match:
+ Event: 'NewConnectedLine'
+ Channel: 'PJSIP/charlie-.*|PJSIP/alice-.*'
+ ChannelStateDesc: 'Up'
+ ConnectedLineNum: 'alice|charlie'
+ ConnectedLineName: 'Alice|Charlie'
+ count: '>2'
+ trigger-on-count: True
+ ami-actions:
+ action:
+ action: 'Hangup'
+ channel: '/^PJSIP/charlie-.*$/'
+
+properties:
+ minversion: '13.8.0'
+ dependencies:
+ - python : twisted
+ - python : starpy
+ - asterisk : app_dial
+ - asterisk : chan_pjsip
+ - asterisk : res_pjsip_caller_id
+ - asterisk : res_pjsip_session
+ tags:
+ - PJSIP
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/tests.yaml b/tests/channels/pjsip/transfers/attended_transfer/nominal/tests.yaml
index b56c877..9056fa9 100644
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/tests.yaml
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/tests.yaml
@@ -3,6 +3,7 @@
- test: 'caller_local'
- test: 'caller_local_blonde'
- test: 'callee_local'
+ - test: 'callee_local_anonymous'
- test: 'callee_local_blonde'
- test: 'caller_remote'
- test: 'callee_remote'
--
To view, visit https://gerrit.asterisk.org/2371
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Gerrit-MessageType: merged
Gerrit-Change-Id: If9ebc98bd60d9ae7d5c90ddc57ae1a3f5b4f8f90
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: George Joseph <george.joseph at fairview5.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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