[Asterisk-code-review] res pjsip: Call Transfer with Empty Extension (testsuite[master])
Torrey Searle
asteriskteam at digium.com
Fri Mar 10 03:07:17 CST 2017
Torrey Searle has uploaded a new change for review. ( 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
Change-Id: Iba651ccc278ebd18ba854b4d85b986ccb7c0e6ba
---
A tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/configs/ast1/extensions.conf
A tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/configs/ast1/pjsip.conf
A tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/run-test
A tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/A_PARTY.xml
A tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/B_PARTY.xml
A tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/C_PARTY.xml
A tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/input_B.csv
A tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/test-config.yaml
M tests/channels/pjsip/transfers/blind_transfer/tests.yaml
9 files changed, 614 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/52/5152/1
diff --git a/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/configs/ast1/extensions.conf b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/configs/ast1/extensions.conf
new file mode 100644
index 0000000..27d761a
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/configs/ast1/extensions.conf
@@ -0,0 +1,37 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+;xfer can have have extensions starting from letters
+exten => s,1,ExecIf($[${SIPTRANSFER} : "yes"]?Set(PEER=sbc)
+exten => s,2,ExecIf($[${SIPTRANSFER} : "yes"]?Set(_SIPHEADER=X-VCC-Destination: ${SIPREFERTOHDR})
+exten => s,3,ExecIf($[${SIPTRANSFER} : "yes"]?goto(callcontrol,s,1) ;default extension when Refer-To doesn't contain any extension
+exten => s,4,Hangup()
+
+; otherwise do normal logic
+exten => _X.,1,Set(PEER=sbc_g729); the init call will switch to the appropriate context if all goes well
+exten => _X.,2,goto(callcontrol,s,1)
+exten => _X.,3,Hangup()
+
+exten => h,1,NoOp(Inside Hangup)
+
+[predial]
+exten => s,1,NoOp(Invoking Pre-Dial)
+exten => s,n,While($["${SET(SIP_HEADER=${POP(SIP_HEADERS,\t)})}" != ""])
+exten => s,n,NoOp(Adding header ${SIP_HEADER})
+exten => s,n,Set(SIP_KEY=${SHIFT(SIP_HEADER,:)})
+exten => s,n,Set(PJSIP_HEADER(add,${SIP_KEY})=${SIP_HEADER:1})
+exten => s,n,EndWhile
+
+[callcontrol]
+exten => s,1,Dial(PJSIP/${PEER},180,b(predial^s^1))
+exten => s,n,Hangup()
+
+exten => h,1,NoOp(Inside hangup)
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/configs/ast1/pjsip.conf b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..9c80f5d
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/configs/ast1/pjsip.conf
@@ -0,0 +1,114 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 5061
+
+[sbc_g729]
+port = 5700
+
+[sbc]
+port = 5800
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+user_agent = Vox Callcontrol
+
+[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:127.0.0.1:5061
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+dtmf_mode = rfc4733
+disallow = all
+allow = alaw
+allow = ulaw
+allow = g729
+allow = h263p
+allow = h264
+direct_media = no
+send_rpid = yes
+sdp_session = session
+dtls_verify = no
+dtls_rekey = 300
+dtls_cert_file = /etc/asterisk/keys/asterisk.crt
+dtls_private_key = /etc/asterisk/keys/asterisk.key
+dtls_cipher = ALL
+aors = PEER_A
+
+[sbc_g729]
+type = aor
+contact = sip:127.0.0.1:5700
+
+[sbc_g729]
+type = endpoint
+context = default
+dtmf_mode = rfc4733
+disallow = all
+allow = alaw
+allow = ulaw
+allow = g729
+allow = h263p
+allow = h264
+direct_media = no
+send_rpid = yes
+sdp_session = session
+dtls_verify = no
+dtls_rekey = 300
+dtls_cert_file = /etc/asterisk/keys/asterisk.crt
+dtls_private_key = /etc/asterisk/keys/asterisk.key
+dtls_cipher = ALL
+aors = sbc_g729
+
+[sbc]
+type = aor
+contact = sip:127.0.0.1:5800
+
+[sbc]
+type = endpoint
+context = default
+dtmf_mode = rfc4733
+disallow = all
+allow = alaw
+allow = ulaw
+allow = g729
+allow = h263p
+allow = h264
+direct_media = no
+send_rpid = yes
+sdp_session = session
+dtls_verify = no
+dtls_rekey = 300
+dtls_cert_file = /etc/asterisk/keys/asterisk.crt
+dtls_private_key = /etc/asterisk/keys/asterisk.key
+dtls_cipher = ALL
+aors = sbc
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/run-test b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/run-test
new file mode 100755
index 0000000..2e196b2
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/run-test
@@ -0,0 +1,90 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2010, Digium, Inc.
+Russell Bryant <russell at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+import logging
+import signal
+import subprocess
+import time
+
+sys.path.append("lib/python")
+sys.path.append("utils")
+
+from twisted.internet import reactor
+from asterisk.sipp import SIPpTest
+
+import voxcallcontrol
+
+logger = logging.getLogger(__name__)
+
+WORKING_DIR = os.path.abspath(os.path.dirname(__file__))
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+e164_B = "3200000000"
+e164_C = "3200000001"
+sippA_logfile = WORKING_DIR + "/A_PARTY.log"
+sippA_errfile = WORKING_DIR + "/A_PARTY_ERR.log"
+sippB_logfile = WORKING_DIR + "/B_PARTY.log"
+sippB_errfile = WORKING_DIR + "/B_PARTY_ERR.log"
+sippC_logfile = WORKING_DIR + "/C_PARTY.log"
+sippC_errfile = WORKING_DIR + "/C_PARTY_ERR.log"
+
+SIPP_SCENARIOS = [
+ {
+ 'scenario' : 'C_PARTY.xml',
+ '-i' : '127.0.0.1',
+ '-p' : '5800',
+ '-mp' : '6400',
+ '-message_file' : sippC_logfile,
+ '-error_file' : sippC_errfile,
+ '-trace_msg' : '-trace_err',
+ },
+ {
+ 'scenario' : 'B_PARTY.xml',
+ '-i' : '127.0.0.1',
+ '-p' : '5700',
+ '-mp' : '6300',
+ '-message_file' : sippB_logfile,
+ '-error_file' : sippB_errfile,
+ '-trace_msg' : '-trace_err',
+ },
+ {
+ 'scenario' : 'A_PARTY.xml',
+ '-i' : '127.0.0.1',
+ '-p' : '5061',
+ '-d' : '15000',
+ '-s' : '3228080970',
+ '-message_file' : sippA_logfile,
+ '-error_file' : sippA_errfile,
+ '-trace_msg' : '-trace_err',
+ },
+]
+
+
+def main():
+
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ test.reactor_timeout = 55;
+
+ time.sleep(10) #Wait 5 seconds to ensure that all the sockets are open before running the test
+
+ reactor.run()
+
+ if not test.passed:
+ return 1
+
+ return 0
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
diff --git a/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/A_PARTY.xml b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/A_PARTY.xml
new file mode 100644
index 0000000..2a7fb7a
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/A_PARTY.xml
@@ -0,0 +1,112 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "bansallaptop.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 'uac' scenario. -->
+<!-- -->
+<scenario name="A_PARTY.xml">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@[remote_ip] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:bansallaptop@[local_ip]:[local_port]>;tag=[call_number]
+ To: bansalphone <sip:[service]@[remote_ip]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:bansallaptop@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=bansallaptop 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>
+
+
+ <!-- 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" rrs="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 [next_url] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: bansallaptop <sip:bansallaptop@[local_ip]:[local_port]>;tag=[call_number]
+ To: bansalphone <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:bansallaptop@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ [routes]
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause/>
+ <send retrans="500">
+ <![CDATA[
+
+ BYE [next_url] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: bansallaptop <sip:bansallaptop@[local_ip]:[local_port]>;tag=[call_number]
+ To: bansalphone <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:bansallaptop@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+ [routes]
+
+ ]]>
+ </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 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/blind_transfer/call_transfer_no_ext/sipp/B_PARTY.xml b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/B_PARTY.xml
new file mode 100644
index 0000000..999b10f
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/B_PARTY.xml
@@ -0,0 +1,153 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="B_PARTY.xml">
+
+<recv request="INVITE" crlf="true" rrs="true">
+<action>
+ <ereg regexp=".*" search_in="hdr" header="From:" check_it="true" assign_to="1"/>
+ <ereg regexp=".*" search_in="hdr" header="To:" check_it="true" assign_to="2"/>
+ <ereg regexp="[[:punct:]](.*)[[:punct:]]" search_in="hdr" header="Contact:" check_it="true" assign_to="6,3" />
+ <log message="Log to avoid the problem of not using $6 [$6]"/>
+</action>
+
+</recv>
+
+
+
+<send>
+<![CDATA[
+
+SIP/2.0 100 Trying
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+<send retrans="500">
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:];tag=[call_number]
+[last_CSeq:]
+[last_Record-Route]
+Contact: <sip:bansallaptop@[local_ip]:[local_port];user=phone>
+Content-Type: application/sdp
+Content-Length: [len]
+
+v=0
+o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+s=Sip Call
+c=IN IP[media_ip_type] [media_ip]
+t=0 0
+m=audio [media_port] RTP/AVP 8
+a=rtpmap:8 PCMA/8000
+
+]]>
+</send>
+
+<recv request="ACK"
+ rtd="true"
+ crlf="true">
+</recv>
+
+<pause milliseconds="5000"/>
+
+<send retrans="500">
+ <![CDATA[
+
+ REFER [$3] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+ CSeq: 1 REFER
+ From: [$2];tag=[call_number]
+ To: [$1]
+ [last_Call-ID:]
+ Max-Forwards: 69
+ User-Agent: Vox Callcontrol
+ Contact: <sip:bansallaptop@[local_ip]:[local_port];user=phone>
+ Refer-To: <sip:[local_ip]:[local_port];transport=tcp;lr>
+ Referred-By: <sip:bansallaptop@[local_ip]:[local_port]>
+ Content-Length: 0
+ ]]>
+</send>
+
+ <recv response="202" optional="true">
+ </recv>
+
+
+<recv request="NOTIFY"
+ rtd="true"
+ crlf="true">
+</recv>
+
+<send>
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+
+
+<recv request="NOTIFY"
+ rtd="true"
+ crlf="true">
+</recv>
+
+<send>
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+
+<send retrans="500">
+ <![CDATA[
+
+ BYE [$3] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+ CSeq: 2 BYE
+ From: [$2];tag=[call_number]
+ To: [$1]
+ [last_Call-ID:]
+ Max-Forwards: 69
+ User-Agent: Vox Callcontrol
+ Contact: <sip:bansallaptop@[local_ip]:[local_port];user=phone>
+ Content-Length: 0
+ ]]>
+</send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+<!-- definition of the response time repartition table (unit is ms) -->
+<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+<!-- definition of the call length repartition table (unit is ms) -->
+<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/C_PARTY.xml b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/C_PARTY.xml
new file mode 100644
index 0000000..b5d28a1
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/C_PARTY.xml
@@ -0,0 +1,92 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Re-Invite problem 1">
+
+<recv request="INVITE" crlf="true" rrs="true">
+<action>
+ <ereg regexp=".*sip:127.0.0.1:.*;transport=tcp;lr$" search_in="hdr" header="X-VCC-Destination:" check_it="true" assign_to="1" />
+ <log message="Log to avoid the problem of not using $1 [$1]"/>
+</action>
+
+</recv>
+
+
+
+<send>
+<![CDATA[
+
+SIP/2.0 100 Trying
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+<send retrans="500">
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:];tag=[call_number]
+[last_CSeq:]
+[last_Record-Route]
+Contact: <sip:bansallaptop@[local_ip]:[local_port];user=phone>
+Content-Type: application/sdp
+Content-Length: [len]
+
+v=0
+o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+s=Sip Call
+c=IN IP[media_ip_type] [media_ip]
+t=0 0
+m=audio [media_port] RTP/AVP 8
+a=rtpmap:8 PCMA/8000
+
+]]>
+</send>
+
+<recv request="ACK"
+ rtd="true"
+ crlf="true">
+</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"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/input_B.csv b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/input_B.csv
new file mode 100644
index 0000000..d31f6d7
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/sipp/input_B.csv
@@ -0,0 +1,2 @@
+SEQUENTIAL
+3200000001
\ No newline at end of file
diff --git a/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/test-config.yaml b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/test-config.yaml
new file mode 100644
index 0000000..91299d6
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/call_transfer_no_ext/test-config.yaml
@@ -0,0 +1,13 @@
+testinfo:
+ summary: 'Test call transfer and switching the transport protocol'
+ description: |
+ 'Test call transfer case. A and B setup a call on UDP, B sends a REFER and indicates transport=tcp in Refer-To header
+ This test case verifies that Asterisk is able to retrieve the transport parameter in Refer-to header and forward it.
+ In this case, C party is emulating the VoxSBC, so it expects X-VCC-Destination header with transport=tcp.'
+
+properties:
+ minversion: '11.0.0'
+ dependencies:
+ - app : 'sipp'
+ tags:
+ - SIP
diff --git a/tests/channels/pjsip/transfers/blind_transfer/tests.yaml b/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
index a529add..50fa4da 100644
--- a/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
+++ b/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
@@ -9,4 +9,5 @@
- test: 'caller_with_hold'
- test: 'caller_with_hold_drop_options'
- test: 'goto_on_blindxfr'
+ - test: 'call_transfer_no_ext'
--
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Gerrit-Change-Id: Iba651ccc278ebd18ba854b4d85b986ccb7c0e6ba
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Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
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