[Asterisk-code-review] pjsip: Add test for 'ip' option of 'identify by' on endpoint. (testsuite[master])
Joshua Colp
asteriskteam at digium.com
Wed Oct 25 05:43:50 CDT 2017
Joshua Colp has uploaded this change for review. ( https://gerrit.asterisk.org/6908
Change subject: pjsip: Add test for 'ip' option of 'identify_by' on endpoint.
......................................................................
pjsip: Add test for 'ip' option of 'identify_by' on endpoint.
This test sends various calls to Asterisk that initially match
based on the username portion of the From header. The endpoint
is configured to not allow it to be matched based on username.
Afterwards matching based on IP address occurs to a valid
endpoint. If calls work then the right endpoint has been
matched. If calls fail then the endpoint was matched based
on username which is incorrect.
ASTERISK-27206
Change-Id: Ieb4e504b0063c723f6cdafd0f78281e33027f7ac
---
A tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/configs/ast1/extensions.conf
A tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/configs/ast1/pjsip.conf
A tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/echo_with_deferred_sdp.xml
A tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/echo_with_initial_sdp.xml
A tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/playback_with_deferred_sdp.xml
A tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/playback_with_initial_sdp.xml
A tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/test-config.yaml
M tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/tests.yaml
M tests/channels/pjsip/dialplan_functions/pjsip_endpoint/configs/ast1/pjsip.conf
9 files changed, 490 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/08/6908/1
diff --git a/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/configs/ast1/extensions.conf b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/configs/ast1/extensions.conf
new file mode 100644
index 0000000..bcea565
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/configs/ast1/extensions.conf
@@ -0,0 +1,16 @@
+[default]
+exten => echo,1,Answer()
+same => n,Echo()
+same => n,Hangup()
+
+exten => playback,1,Answer()
+same => n,Playback(hello-world)
+same => n,Hangup()
+
+exten => early,1,Progress()
+same => n,Playback(hello-world,noanswer)
+same => n,Hangup(INTERWORKING)
+
+;This dialstring can be altered once endpoints can be used directly
+exten => bob,1,Dial(PJSIP/sip:bob at 127.0.0.1:5062)
+same => n,Hangup()
diff --git a/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/configs/ast1/pjsip.conf b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..d3cd18a
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/configs/ast1/pjsip.conf
@@ -0,0 +1,61 @@
+[global]
+type=global
+endpoint_identifier_order=username,ip
+
+[local-transport-template](!)
+type=transport
+bind=127.0.0.1
+
+[local-transport6-template](!)
+type=transport
+bind=[::1]
+
+[local-transport-udp](local-transport-template)
+protocol=udp
+
+[local-transport-udp6](local-transport6-template)
+protocol=udp
+
+[local-transport-tcp](local-transport-template)
+protocol=tcp
+
+[local-transport-tcp6](local-transport6-template)
+protocol=tcp
+
+[endpoint-template-ipv4](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+media_address=127.0.0.1
+identify_by=ip
+
+[endpoint-template-ipv6](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+media_address=[::1]
+rtp_ipv6=yes
+identify_by=ip
+
+[alice-ipv4-udp](endpoint-template-ipv4)
+
+[alice-ipv4-tcp](endpoint-template-ipv4)
+
+[alice-ipv6-udp](endpoint-template-ipv6)
+
+[alice-ipv6-tcp](endpoint-template-ipv6)
+
+[unknown](endpoint-template-ipv4)
+context=does-not-exist
+
+[identify-template](!)
+type=identify
+
+[alice-identify-ipv4](identify-template)
+endpoint=alice-ipv4-udp
+match=127.0.0.1
+
+[alice-identify-ipv6](identify-template)
+endpoint=alice-ipv6-udp
+match=[::1]
+
diff --git a/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/echo_with_deferred_sdp.xml b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/echo_with_deferred_sdp.xml
new file mode 100644
index 0000000..5aa6143
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/echo_with_deferred_sdp.xml
@@ -0,0 +1,84 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in ACK">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: 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/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/echo_with_initial_sdp.xml b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/echo_with_initial_sdp.xml
new file mode 100644
index 0000000..2608c36
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/echo_with_initial_sdp.xml
@@ -0,0 +1,85 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: 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/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/playback_with_deferred_sdp.xml b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/playback_with_deferred_sdp.xml
new file mode 100644
index 0000000..29fda2f
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/playback_with_deferred_sdp.xml
@@ -0,0 +1,80 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to playback with SDP in ACK">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <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>
+
+ <!-- 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/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/playback_with_initial_sdp.xml b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/playback_with_initial_sdp.xml
new file mode 100644
index 0000000..ea7d68a
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/sipp/playback_with_initial_sdp.xml
@@ -0,0 +1,81 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to playback with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:unknown@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <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>
+
+ <!-- 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/basic_calls/incoming/nominal/unauthed/ident_by_host_only/test-config.yaml b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/test-config.yaml
new file mode 100644
index 0000000..e2bd547
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host_only/test-config.yaml
@@ -0,0 +1,81 @@
+testinfo:
+ summary: 'Tests incoming calls without authentication matching only on IP address'
+ description: |
+ 'Run a SIPp scenario that sends various calls to res_pjsip, which should be matched only on IP address'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ reactor-timeout: 80
+ fail-on-any: False
+ test-iterations:
+ # IPv4 & UDP
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'playback_with_initial_sdp.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice-ipv4-udp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'echo_with_initial_sdp.xml', '-i': '127.0.0.1', '-p': '5061', '-d': '5000', '-s': 'alice-ipv4-udp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'playback_with_deferred_sdp.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice-ipv4-udp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'echo_with_deferred_sdp.xml', '-i': '127.0.0.1', '-p': '5061', '-d': '5000', '-s': 'alice-ipv4-udp'} }
+
+ # IPv4 & TCP
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'playback_with_initial_sdp.xml', '-i': '127.0.0.1', '-p': '5061', '-t': 't1', '-s': 'alice-ipv4-tcp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'echo_with_initial_sdp.xml', '-i': '127.0.0.1', '-p': '5062', '-t': 't1', '-d': '5000', '-s': 'alice-ipv4-tcp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'playback_with_deferred_sdp.xml', '-i': '127.0.0.1', '-p': '5066', '-t': 't1', '-s': 'alice-ipv4-tcp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'echo_with_deferred_sdp.xml', '-i': '127.0.0.1', '-p': '5067', '-t': 't1', '-d': '5000', '-s': 'alice-ipv4-tcp'} }
+
+ # IPv6 & UDP
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'playback_with_initial_sdp.xml', '-i': '[::1]', '-p': '5061', '-s': 'alice-ipv6-udp'} }
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'echo_with_initial_sdp.xml', '-i': '[::1]', '-p': '5062', '-d': '5000', '-s': 'alice-ipv6-udp'} }
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'playback_with_deferred_sdp.xml', '-i': '[::1]', '-p': '5066', '-s': 'alice-ipv6-udp'} }
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'echo_with_deferred_sdp.xml', '-i': '[::1]', '-p': '5067', '-d': '5000', '-s': 'alice-ipv6-udp'} }
+
+ # IPv6 & TCP
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'playback_with_initial_sdp.xml', '-i': '[::1]', '-p': '5061', '-t': 't1', '-s': 'alice-ipv6-tcp'} }
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'echo_with_initial_sdp.xml', '-i': '[::1]', '-p': '5062', '-t': 't1', '-d': '5000', '-s': 'alice-ipv6-tcp'} }
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'playback_with_deferred_sdp.xml', '-i': '[::1]', '-p': '5066', '-t': 't1', '-s': 'alice-ipv6-tcp'} }
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'echo_with_deferred_sdp.xml', '-i': '[::1]', '-p': '5067', '-t': 't1', '-d': '5000', '-s': 'alice-ipv6-tcp'} }
+
+properties:
+ minversion: [ '13.19.0', '15.2.0' ]
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'res_pjsip'
+ - asterisk : 'res_pjsip_session'
+ - asterisk : 'chan_pjsip'
+ - asterisk : 'res_pjsip_endpoint_identifier_ip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/tests.yaml b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/tests.yaml
index ed1244b..68e17f6 100644
--- a/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/tests.yaml
+++ b/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/tests.yaml
@@ -1,5 +1,6 @@
tests:
- test: 'ident_by_host'
+ - test: 'ident_by_host_only'
- test: 'ident_by_user'
- test: 'ident_by_user_drop_options'
- test: 'ident_by_header'
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_endpoint/configs/ast1/pjsip.conf b/tests/channels/pjsip/dialplan_functions/pjsip_endpoint/configs/ast1/pjsip.conf
index 9cb6dbb..3baa86b 100644
--- a/tests/channels/pjsip/dialplan_functions/pjsip_endpoint/configs/ast1/pjsip.conf
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_endpoint/configs/ast1/pjsip.conf
@@ -9,6 +9,7 @@
type=endpoint
context=default
allow=!all,ulaw,alaw
+identify_by=username
[aors-template](!)
type=aor
--
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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: Ieb4e504b0063c723f6cdafd0f78281e33027f7ac
Gerrit-Change-Number: 6908
Gerrit-PatchSet: 1
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
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