[Asterisk-code-review] testsuite: Move pjsip/transfers/../callee local anonymous t... (testsuite[master])
George Joseph
asteriskteam at digium.com
Wed May 10 11:01:49 CDT 2017
George Joseph has uploaded a new change for review. ( https://gerrit.asterisk.org/5613 )
Change subject: testsuite: Move pjsip/transfers/../callee_local_anonymous to headers
......................................................................
testsuite: Move pjsip/transfers/../callee_local_anonymous to headers
We've determined that anonymizing the From header when doing a
callee transfer re-invite is not correct. There are no other
anonymized From tests though so this one was moved to the "headers"
directory and converted to do just a basic call with
callerid_privacy=prohib set. The UAS (Bob) verifies that the From
header on the INVITE is anonymized.
A second test was also added to confirm that when
callerid_privacy=prohib is NOT set, the From header isn't
anonymized.
Change-Id: I5b67ff0b7b1909b1db2edcd2f982a6a057c8c12e
---
A tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/extensions.conf
R tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/pjsip.conf
A tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/alice.xml
A tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/bob.xml
A tests/channels/pjsip/headers/anonymous_from_basic_call/test-config.yaml
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/extensions.conf
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/pjsip.conf
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/alice.xml
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/bob.xml
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/test-config.yaml
A tests/channels/pjsip/headers/tests.yaml
M tests/channels/pjsip/tests.yaml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml
18 files changed, 443 insertions(+), 785 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/13/5613/1
diff --git a/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/extensions.conf b/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/extensions.conf
new file mode 100644
index 0000000..bc9969a
--- /dev/null
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/extensions.conf
@@ -0,0 +1,5 @@
+
+[default]
+exten => bob,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/headers/anonymous_from_basic_call/configs/ast1/pjsip.conf
similarity index 64%
rename from tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf
rename to tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/pjsip.conf
index d7e4874..1b000aa 100644
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/pjsip.conf
@@ -24,15 +24,5 @@
[bob]
type=aor
-contact=sip:bob at 127.0.0.1:5066
+contact=sip:bob at 127.0.0.1:5063
-[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/headers/anonymous_from_basic_call/sipp/alice.xml b/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/alice.xml
new file mode 100644
index 0000000..976cbec
--- /dev/null
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/alice.xml
@@ -0,0 +1,79 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAC Requestor">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:bob@[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="180" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:bob@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:bob@[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>
+
+ <pause milliseconds="1000"/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/bob.xml b/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/bob.xml
new file mode 100644
index 0000000..1fffcf1
--- /dev/null
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/bob.xml
@@ -0,0 +1,91 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+
+ <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/2.0 180 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">
+ </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>
+
+ <Reference variables="asserted_identity" />
+ <Reference variables="remote_party_id" />
+ <Reference variables="from" />
+
+</scenario>
+
diff --git a/tests/channels/pjsip/headers/anonymous_from_basic_call/test-config.yaml b/tests/channels/pjsip/headers/anonymous_from_basic_call/test-config.yaml
new file mode 100644
index 0000000..12fe669
--- /dev/null
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/test-config.yaml
@@ -0,0 +1,30 @@
+testinfo:
+ summary: Test anonymized From headers.
+ description: |
+ Alice calls Bob with callerid_privacy=prohib
+ Bob verifies that the From header is anonymized.
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: sipp.SIPpTestCase
+
+test-object-config:
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario':'alice.xml', '-p':'5062'} }
+ - { 'key-args': {'scenario':'bob.xml', '-p':'5063'} }
+
+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/headers/non-anonymous_from_basic_call/configs/ast1/extensions.conf b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/extensions.conf
new file mode 100644
index 0000000..bc9969a
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/extensions.conf
@@ -0,0 +1,5 @@
+
+[default]
+exten => bob,1,NoOp()
+ same => n,Dial(PJSIP/bob)
+ same => n,Hangup()
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/pjsip.conf b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..9e29cca
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/pjsip.conf
@@ -0,0 +1,27 @@
+[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
+
+[alice](endpoint)
+callerid=Alice <alice>
+
+[bob](endpoint)
+aors=bob
+callerid=Bob <bob>
+
+[bob]
+type=aor
+contact=sip:bob at 127.0.0.1:5063
+
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/alice.xml b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/alice.xml
new file mode 100644
index 0000000..976cbec
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/alice.xml
@@ -0,0 +1,79 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAC Requestor">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:bob@[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="180" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:bob@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:bob@[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>
+
+ <pause milliseconds="1000"/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/bob.xml b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/bob.xml
new file mode 100644
index 0000000..c05dfab
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/bob.xml
@@ -0,0 +1,91 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+
+ <recv request="INVITE">
+ <action>
+ <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
+ 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/2.0 180 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">
+ </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>
+
+ <Reference variables="asserted_identity" />
+ <Reference variables="remote_party_id" />
+ <Reference variables="from" />
+
+</scenario>
+
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/test-config.yaml b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/test-config.yaml
new file mode 100644
index 0000000..132dbe5
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/test-config.yaml
@@ -0,0 +1,30 @@
+testinfo:
+ summary: Test non-anonymized From headers.
+ description: |
+ Alice calls Bob without callerid_privacy=prohib
+ Bob verifies that the From header is NOT anonymized.
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: sipp.SIPpTestCase
+
+test-object-config:
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario':'alice.xml', '-p':'5062'} }
+ - { 'key-args': {'scenario':'bob.xml', '-p':'5063'} }
+
+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/headers/tests.yaml b/tests/channels/pjsip/headers/tests.yaml
new file mode 100644
index 0000000..7c3b64d
--- /dev/null
+++ b/tests/channels/pjsip/headers/tests.yaml
@@ -0,0 +1,4 @@
+# Enter tests here in the order they should be considered for execution:
+tests:
+ - test: 'anonymous_from_basic_call'
+ - test: 'non-anonymous_from_basic_call'
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index 64b3ec0..6c69ab1 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -6,6 +6,7 @@
- dir: 'configuration'
- dir: 'dialplan_functions'
- dir: 'diversion'
+ - dir: 'headers'
- dir: 'identify'
- dir: 'message'
- dir: 'nat'
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
deleted file mode 100644
index a299118..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf
+++ /dev/null
@@ -1,9 +0,0 @@
-
-[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/sipp/referee.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
deleted file mode 100644
index f4166e1..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
+++ /dev/null
@@ -1,137 +0,0 @@
-<?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
deleted file mode 100644
index 950cc54..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml
+++ /dev/null
@@ -1,252 +0,0 @@
-<?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
deleted file mode 100644
index 321e53f..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml
+++ /dev/null
@@ -1,141 +0,0 @@
-<?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
deleted file mode 100644
index 69c014d..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml
+++ /dev/null
@@ -1,166 +0,0 @@
-<?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
deleted file mode 100644
index 160148b..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml
+++ /dev/null
@@ -1,69 +0,0 @@
-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
--
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I5b67ff0b7b1909b1db2edcd2f982a6a057c8c12e
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: George Joseph <gjoseph at digium.com>
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