[Asterisk-code-review] testsuite: SIP route header is missing on UPDATE (testsuite[18])
Friendly Automation
asteriskteam at digium.com
Wed Apr 27 03:39:45 CDT 2022
Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/testsuite/+/18461 )
Change subject: testsuite: SIP route header is missing on UPDATE
......................................................................
testsuite: SIP route header is missing on UPDATE
Add tests to ensure we receive Route header in UPDATE
ASTERISK-29955
Change-Id: I55caad3cb2a156b8e3f2f24dd10db5ebe67910d2
---
A tests/channels/SIP/sip_semi_attended_transfer_record_route/configs/ast1/extensions.conf
A tests/channels/SIP/sip_semi_attended_transfer_record_route/configs/ast1/sip.conf
A tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/referee.xml
A tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/referer_uas.xml
A tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/uac-no-hangup.xml
A tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/uas.xml
A tests/channels/SIP/sip_semi_attended_transfer_record_route/test-config.yaml
M tests/channels/SIP/tests.yaml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/configs/ast1/extensions.conf
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/configs/ast1/pjsip.conf
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/referee.xml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/referer_uas.xml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/uac-no-hangup.xml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/uas.xml
A tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/test-config.yaml
M tests/channels/pjsip/transfers/attended_transfer/nominal/tests.yaml
16 files changed, 1,560 insertions(+), 0 deletions(-)
Approvals:
George Joseph: Looks good to me, approved
Friendly Automation: Approved for Submit
diff --git a/tests/channels/SIP/sip_semi_attended_transfer_record_route/configs/ast1/extensions.conf b/tests/channels/SIP/sip_semi_attended_transfer_record_route/configs/ast1/extensions.conf
new file mode 100644
index 0000000..b66e24e
--- /dev/null
+++ b/tests/channels/SIP/sip_semi_attended_transfer_record_route/configs/ast1/extensions.conf
@@ -0,0 +1,8 @@
+[default]
+exten => call_c,1,NoOp()
+ same => n,Dial(SIP/charlie)
+ same => n,Hangup()
+
+exten => alice,1,NoOp()
+ same => n,Dial(SIP/bob)
+ same => n,Hangup()
diff --git a/tests/channels/SIP/sip_semi_attended_transfer_record_route/configs/ast1/sip.conf b/tests/channels/SIP/sip_semi_attended_transfer_record_route/configs/ast1/sip.conf
new file mode 100644
index 0000000..ffd805e
--- /dev/null
+++ b/tests/channels/SIP/sip_semi_attended_transfer_record_route/configs/ast1/sip.conf
@@ -0,0 +1,36 @@
+[general]
+canreinvite=no
+sipdebug=yes
+udpbindaddr=0.0.0.0
+
+rpid_update=yes
+trustrpid=yes
+sendrpid=pai
+
+
+[alice]
+insecure=invite
+host=127.0.0.1
+port=5068
+context=default
+type=friend
+
+[bob]
+insecure=invite
+host=127.0.0.1
+port=5066
+context=default
+type=friend
+
+[charlie]
+insecure=invite
+host=127.0.0.1
+port=5067
+context=default
+type=friend
+
+[david]
+insecure=invite
+host=dynamic
+context=default
+type=friend
diff --git a/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/referee.xml b/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/referee.xml
new file mode 100644
index 0000000..cdb3284
--- /dev/null
+++ b/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/referee.xml
@@ -0,0 +1,120 @@
+<?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]
+ Record-Route: <sip:[local_ip]:[local_port];transport=[transport];lr>
+ From: <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:transfer@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Max-Forwards: 70
+ X-SIPP: referee.xml
+ 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">
+ <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" />
+
+ <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 response="603" />
+
+ <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>
+
+ <!-- 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/SIP/sip_semi_attended_transfer_record_route/sipp/referer_uas.xml b/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/referer_uas.xml
new file mode 100644
index 0000000..888d4d0
--- /dev/null
+++ b/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/referer_uas.xml
@@ -0,0 +1,221 @@
+<?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" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ Record-Route: <sip:[local_ip]:[local_port];transport=[transport];lr>
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ X-SIPP: referer_uas.xml
+ 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">
+ <action>
+ <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: [$outbound_to_header]
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Max-Forwards: 70
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv
+ response="100"
+ optional="true" />
+ <recv
+ response="101"
+ optional="true" />
+ <recv
+ response="180"
+ optional="true" />
+ <recv
+ response="200"
+ rtd="true" />
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[local_ip]:[local_port] SIP/2.0
+ [last_Via]
+ [last_From]
+ [last_To]
+ Call-ID: [call_id]
+ CSeq: [cseq] ACK
+ Contact: sip:bob@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <sendCmd>
+ <![CDATA[
+ Call-ID: REMOTE[call_id]
+ Start the Echo Leg
+ ]]>
+ </sendCmd>
+
+ <recvCmd>
+ <action>
+ <ereg
+ regexp=" (.+)"
+ search_in="hdr"
+ header="Remote-URI:"
+ check_it="true"
+ assign_to="1,remote_contact" />
+ <ereg
+ regexp=" (.+)"
+ search_in="hdr"
+ header="Remote-To-Tag:"
+ check_it="true"
+ assign_to="2,remote_to_tag" />
+ <ereg
+ regexp=" (.+)"
+ search_in="hdr"
+ header="Remote-From-Tag:"
+ check_it="true"
+ assign_to="3,remote_from_tag" />
+ </action>
+ </recvCmd>
+ <Reference variables="1,2,3" />
+
+ <send>
+ <![CDATA[
+
+ REFER sip:call_c@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ [last_From:]
+ [last_To]
+ [last_Call-ID:]
+ CSeq: [cseq] REFER
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Max-Forwards: 70
+ Refer-to: <[$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" />
+
+ <recv request="NOTIFY" />
+ <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>
+
+
+ <pause milliseconds="5000" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE 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] BYE
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Max-Forwards: 70
+ 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/SIP/sip_semi_attended_transfer_record_route/sipp/uac-no-hangup.xml b/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/uac-no-hangup.xml
new file mode 100644
index 0000000..d678884
--- /dev/null
+++ b/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/uac-no-hangup.xml
@@ -0,0 +1,110 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+
+ <nop hide="true">
+ <action>
+ <assignstr
+ assign_to="rr_out"
+ value="<sip:[local_ip]:[local_port];transport=[transport];lr>" />
+ </action>
+ </nop>
+
+ <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]
+ Record-Route: [$rr_out]
+ 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
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ X-SIPP: uac-no-hangup.xml
+ 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="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"/>
+
+</scenario>
+
diff --git a/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/uas.xml b/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/uas.xml
new file mode 100644
index 0000000..4899731
--- /dev/null
+++ b/tests/channels/SIP/sip_semi_attended_transfer_record_route/sipp/uas.xml
@@ -0,0 +1,218 @@
+<?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 -->
+<!-- MERCHANTARouteTNESS 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">
+
+ <nop hide="true">
+ <action>
+ <assignstr
+ assign_to="rr_out"
+ value="<sip:[local_ip]:[local_port];transport=[transport];lr>" />
+ </action>
+ </nop>
+
+
+<!-- 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">
+
+ <action>
+ <ereg
+ regexp=".*"
+ header="Via:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="original_via"/>
+ </action>
+
+ </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/2.0 100 Trying
+ [last_Via:]
+ Record-Route: [$rr_out]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ X-SIPP: uas.xml
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ Record-Route: [$rr_out]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ X-SIPP: uas.xml
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="UPDATE" rtd="true">
+ <action>
+ <ereg
+ regexp="\s*(.*)\s*"
+ search_in="hdr"
+ header="Route:"
+ assign_to="1,r_got"/>
+ <strcmp
+ assign_to="1"
+ variable="r_got"
+ variable2="rr_out" />
+ <test
+ assign_to="r_wrong"
+ variable="1"
+ compare="not_equal"
+ value="" />
+ </action>
+ </recv>
+
+ <nop condexec="r_wrong">
+ <action>
+ <error message="UPDATE Route expected '[$rr_out]' but got '[$r_got]'" />
+ </action>
+ </nop>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ X-SIPP: uas.xml ACK UPDATE
+ Content-Length:0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="200" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ Via: [$original_via]
+ Record-Route: <sip:[local_ip]:[local_port];transport=[transport];lr>
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ CSeq: [cseq-1] INVITE
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ X-SIPP: uas.xml ANSWER CALL
+ 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">
+ <action>
+ <!-- Save the From tag. We'll need it when we send our BYE -->
+ <ereg
+ regexp="(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ <!-- Save the From user portion of URI. We'll need it when we send our BYE -->
+ <ereg
+ regexp="(sip:bob)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_user"/>
+ </action>
+ </recv>
+
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE [$remote_user]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:charlie@[local_ip]:[local_port]>;tag=[call_number]
+ To: <[$remote_user]@[remote_ip]:[remote_port]>[$remote_tag]
+ Call-ID: [call_id]
+ CSeq: [cseq] BYE
+ Contact: sip:charlie@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv
+ request="ACK"
+ optional="true"
+ rtd="true">
+ </recv>
+
+ <recv response="200">
+ </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/SIP/sip_semi_attended_transfer_record_route/test-config.yaml b/tests/channels/SIP/sip_semi_attended_transfer_record_route/test-config.yaml
new file mode 100644
index 0000000..28aef39
--- /dev/null
+++ b/tests/channels/SIP/sip_semi_attended_transfer_record_route/test-config.yaml
@@ -0,0 +1,58 @@
+testinfo:
+ summary: Test performing a callee-initiated semi attended transfer with record-route header via chan_pjsip.
+ 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.
+ Before SIPp #4 answers SIPp #2 initiates an attended transfer via REFER with Replaces information from SIPp #3.
+ SIPp #3 is hung up.
+ SIPp #2 hangs up.
+ SIPp #4 continues to ring until it answers.
+ SIPp #1 receives a connected line update and the values are checked.
+ SIPp #4 answers.
+ SIPp #1 and SIPp #4 are bridged.
+ SIPp #4 receives a connected line update and the values are checked.
+ SIPp #4 hangs up.
+ SIPp #1 is hung up."
+
+test-modules:
+ add-test-to-search-path: True
+ test-object:
+ config-section: test-object-config
+ typename: sipp.SIPpTestCase
+ modules:
+ -
+ config-section: ami-config
+ typename: 'ami.AMIEventModule'
+
+test-object-config:
+ memcheck-delay-stop: 7
+ 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:
+ -
+ type: 'headermatch'
+ conditions:
+ match:
+ Event: 'AttendedTransfer'
+ Result: 'Success'
+ count: 1
+
+properties:
+ dependencies:
+ - python : twisted
+ - python : starpy
+ - asterisk : app_dial
+ - asterisk : chan_sip
+ tags:
+ - SIP
+ - transfer
+
diff --git a/tests/channels/SIP/tests.yaml b/tests/channels/SIP/tests.yaml
index f71357f..b25f4f9 100644
--- a/tests/channels/SIP/tests.yaml
+++ b/tests/channels/SIP/tests.yaml
@@ -67,6 +67,7 @@
- test: 'sip_outbound_proxy'
- test: 'sip_register'
- test: 'sip_register_domain_acl'
+ - test: 'sip_semi_attended_transfer_record_route'
- test: 'sip_tls_call'
- test: 'sip_tls_register'
- test: 'sip_unregister'
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/configs/ast1/extensions.conf b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/configs/ast1/extensions.conf
new file mode 100644
index 0000000..5dc3863
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/configs/ast1/extensions.conf
@@ -0,0 +1,8 @@
+[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_semi_attended_transfer_record_route/configs/ast1/pjsip.conf b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..e5eccf8
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/configs/ast1/pjsip.conf
@@ -0,0 +1,40 @@
+[system]
+type=system
+timer_t1=100
+timer_b=6400
+
+[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
+
+[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_semi_attended_transfer_record_route/sipp/referee.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/referee.xml
new file mode 100644
index 0000000..773f6bd
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/referee.xml
@@ -0,0 +1,120 @@
+<?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]
+ Record-Route: <sip:[local_ip]:[local_port];transport=[transport];lr>
+ From: <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:transfer@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ 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">
+ <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" />
+
+ <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 response="603" />
+
+ <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>
+
+ <!-- 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_semi_attended_transfer_record_route/sipp/referer_uas.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/referer_uas.xml
new file mode 100644
index 0000000..b6f3419
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/referer_uas.xml
@@ -0,0 +1,234 @@
+<?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" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ Record-Route: <sip:[local_ip]:[local_port];transport=[transport];lr>
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ 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">
+ <action>
+ <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
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ 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" />
+
+ <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[$remote_from_tag]>
+ Referred-By: sip:bob@[local_ip]
+ Content-Length: 0
+
+ ]]>
+ </send>
+ <recv
+ response="202"
+ rtd="true" />
+
+ <recv request="NOTIFY" />
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Content-Length:0
+
+ ]]>
+ </send>
+
+ <recv request="NOTIFY" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ Content-Length:0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];rport;received=127.0.0.1;branch=[branch]
+ From: [$outbound_from_header]
+ To: [$outbound_to_header]
+ Call-ID: [call_id]
+ CSeq: [cseq] BYE
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/uac-no-hangup.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/uac-no-hangup.xml
new file mode 100644
index 0000000..dc8b57e
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/uac-no-hangup.xml
@@ -0,0 +1,128 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+
+ <nop hide="true">
+ <action>
+ <assignstr
+ assign_to="rr_out"
+ value="<sip:[local_ip]:[local_port];transport=[transport];lr>" />
+ </action>
+ </nop>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Record-Route: [$rr_out]
+ 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
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ 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="UPDATE">
+ </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]
+
+ ]]>
+ </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>
+
+ <timewait milliseconds="4000"/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/uas.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/uas.xml
new file mode 100644
index 0000000..54fc4b9
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/sipp/uas.xml
@@ -0,0 +1,201 @@
+<?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 -->
+<!-- MERCHANTARouteTNESS 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">
+
+ <nop hide="true">
+ <action>
+ <assignstr
+ assign_to="rr_out"
+ value="<sip:[local_ip]:[local_port];transport=[transport];lr>" />
+ </action>
+ </nop>
+
+
+<!-- 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">
+
+ <action>
+ <ereg
+ regexp=".*"
+ header="Via:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="original_via"/>
+ </action>
+
+ </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:]
+ Record-Route: [$rr_out]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ X-SIPP: uas.xml
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="UPDATE" rtd="true">
+ <action>
+ <ereg
+ regexp="\s*(.*)\s*"
+ search_in="hdr"
+ header="Route:"
+ assign_to="1,r_got"/>
+ <strcmp
+ assign_to="1"
+ variable="r_got"
+ variable2="rr_out" />
+ <test
+ assign_to="r_wrong"
+ variable="1"
+ compare="not_equal"
+ value="" />
+ </action>
+ </recv>
+
+ <nop condexec="r_wrong">
+ <action>
+ <error message="ACK Route expected '[$rr_out]' but got '[$r_got]'" />
+ </action>
+ </nop>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:bob@[local_ip]:[local_port]>
+ X-SIPP: uas.xml ACK UPDATE
+ Content-Length:0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="200" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ Via: [$original_via]
+ Record-Route: <sip:[local_ip]:[local_port];transport=[transport];lr>
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ CSeq: [cseq-1] INVITE
+ Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ X-SIPP: uas.xml ANSWER CALL
+ 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">
+ <action>
+ <!-- Save the From tag. We'll need it when we send our BYE -->
+ <ereg
+ regexp="(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ <!-- Save the From user portion of URI. We'll need it when we send our BYE -->
+ <ereg
+ regexp="(sip:bob)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_user"/>
+ </action>
+ </recv>
+
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE [$remote_user]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:charlie@[local_ip]:[local_port]>;tag=[call_number]
+ To: <[$remote_user]@[remote_ip]:[remote_port]>[$remote_tag]
+ Call-ID: [call_id]
+ CSeq: [cseq] BYE
+ Contact: sip:charlie@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv
+ request="ACK"
+ optional="true"
+ rtd="true">
+ </recv>
+
+ <recv response="200">
+ </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/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/test-config.yaml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/test-config.yaml
new file mode 100644
index 0000000..7e77de8
--- /dev/null
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/test-config.yaml
@@ -0,0 +1,56 @@
+testinfo:
+ summary: Test performing a callee-initiated semi attended transfer with record-route header via chan_pjsip.
+ 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.
+ Before SIPp #4 answers SIPp #2 initiates an attended transfer via REFER with Replaces information from SIPp #3.
+ SIPp #3 is hung up.
+ SIPp #2 hangs up.
+ SIPp #4 continues to ring until it answers.
+ SIPp #1 receives a connected line update and the values are checked.
+ SIPp #4 answers.
+ SIPp #1 and SIPp #4 are bridged.
+ SIPp #4 receives a connected line update and the values are checked.
+ SIPp #4 hangs up.
+ SIPp #1 is hung up."
+
+test-modules:
+ add-test-to-search-path: True
+ test-object:
+ config-section: test-object-config
+ typename: sipp.SIPpTestCase
+ modules:
+ -
+ config-section: ami-config
+ typename: 'ami.AMIEventModule'
+
+test-object-config:
+ memcheck-delay-stop: 7
+ 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:
+ -
+ type: 'headermatch'
+ conditions:
+ match:
+ Event: 'AttendedTransfer'
+ Result: 'Success'
+ count: 1
+
+properties:
+ dependencies:
+ - python : twisted
+ - python : starpy
+ - asterisk : app_dial
+ - asterisk : chan_pjsip
+ 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..0eefcbb 100644
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/tests.yaml
+++ b/tests/channels/pjsip/transfers/attended_transfer/nominal/tests.yaml
@@ -10,3 +10,4 @@
- test: 'callee_local_app'
- test: 'caller_local_direct_media'
- test: 'callee_local_direct_media'
+ - test: 'callee_local_semi_attended_transfer_record_route'
--
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Gerrit-Project: testsuite
Gerrit-Branch: 18
Gerrit-Change-Id: I55caad3cb2a156b8e3f2f24dd10db5ebe67910d2
Gerrit-Change-Number: 18461
Gerrit-PatchSet: 1
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-CC: Mark Petersen <bugs.digium.com at zombie.dk>
Gerrit-MessageType: merged
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