[asterisk-commits] jrose: testsuite/asterisk/trunk r4213 - in /asterisk/trunk/tests/fax: ./ dire...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Sep 23 12:02:35 CDT 2013
Author: jrose
Date: Mon Sep 23 12:02:32 2013
New Revision: 4213
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4213
Log:
testsuite: Add a test for reinvite to T.38 with directmedia
This is a test that sets up a SIP call from one SIPp endpoint to
another with directmedia with just audio and then a reinvite
is issued from one of the peers to the other to change the call
into a T.38 fax session. The test ensures that at this point,
media is routed back through Asterisk.
(closes issue ASTERISK-17273)
Reported by: dario
Review: https://reviewboard.asterisk.org/r/2854/
Added:
asterisk/trunk/tests/fax/directmedia_reinvite_t38/
asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/
asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/
asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/
asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_A.xml (with props)
asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_B.xml (with props)
asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/inject_bridge.csv (with props)
asterisk/trunk/tests/fax/directmedia_reinvite_t38/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/fax/tests.yaml
Added: asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/extensions.conf?view=auto&rev=4213
==============================================================================
--- asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/extensions.conf Mon Sep 23 12:02:32 2013
@@ -1,0 +1,14 @@
+[general]
+PHONE_TO_DIAL=SIP/endpoint_B
+
+[default]
+exten => bypassbridge,1,NoOp()
+ same => n,Dial(SIP/endpoint_B,,g)
+ same => n,UserEvent(TestStatus, extension: bypassbridge)
+ same => n,Hangup()
+
+; Dial with no options; use bridge set up based on peer definitions
+exten => basicdial,1,NoOp()
+ same => n,Dial(SIP/endpoint_B,,g)
+ same => n,UserEvent(TestStatus, extension: basicdial)
+ same => n,Hangup()
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/extensions.conf
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svn:eol-style = native
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/extensions.conf
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/extensions.conf
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/sip.conf?view=auto&rev=4213
==============================================================================
--- asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/sip.conf Mon Sep 23 12:02:32 2013
@@ -1,0 +1,25 @@
+[general]
+allowguest=no
+bindaddr=127.0.0.1
+sipdebug=yes
+context=default
+
+[endpoint_A]
+type=friend
+host=127.0.0.2
+qualify=no
+disallow=all
+allow=ulaw
+insecure=invite
+t38pt_udptl=yes
+directmedia=yes
+
+[endpoint_B]
+type=friend
+t38pt_udptl=yes
+host=127.0.0.3
+qualify=no
+disallow=all
+allow=ulaw
+insecure=invite
+directmedia=yes
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/sip.conf
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svn:eol-style = native
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/sip.conf
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/configs/ast1/sip.conf
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_A.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_A.xml?view=auto&rev=4213
==============================================================================
--- asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_A.xml (added)
+++ asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_A.xml Mon Sep 23 12:02:32 2013
@@ -1,0 +1,218 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone A Hold with IP and Media Restrictions">
+
+ <!-- Initial invite - Call phone B -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:[field2]@[remote_ip]:[remote_port];user=phone>
+ CSeq: 1 INVITE
+ Call-ID: [call_id]
+ Contact: <sip:[field0]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendrecv
+ m=audio 2226 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="180" optional="true" />
+
+ <recv response="183" optional="true" />
+
+ <recv response="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field0] <sip:[field0]@[remote_ip]>;tag=[call_number]
+ To: <sip:[field1]@[remote_ip];user=phone>[peer_tag_param]
+ CSeq: 1 ACK
+ Call-ID: [call_id]
+ Contact: <sip:[field0]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Reinvite to establish directmedia - media flows between Phone A and Phone B -->
+ <recv request="INVITE">
+ <action>
+ <ereg regexp="c=IN IP4 127.0.0.3" search_in="body" check_it="true" assign_to="1" />
+ <log message="Side A - Contact SDP for directmedia reinvite matches expectations: [$1]." />
+ <strcmp assign_to="result" variable="1" value="c=IN IP4 127.0.0.3" />
+ <test assign_to="emptyinv1" variable="result" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="emptyinv1">
+ <action>
+ <error message="Side A - Contact SDP for directmedia reinvite did not match - expected 'c=IN IP4 127.0.0.3' but got [$1]" />
+ </action>
+ </nop>
+
+ <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:[field0]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Testsuite-Track-Phone-A: 1
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendrecv
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+
+ <!-- Reinvite received for T38 - media flows between Enpoint A and Asterisk -->
+ <recv request="INVITE">
+ <action>
+ <ereg regexp="c=IN IP4 127.0.0.1" search_in="body" check_it="true" assign_to="1" />
+ <log message="Side A - Contact SDP for T38 reinvite matches expectations: [$1]." />
+ <strcmp assign_to="result" variable="1" value="c=IN IP4 127.0.0.1" />
+ <test assign_to="emptyinv2" variable="result" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="emptyinv2">
+ <action>
+ <error message="Side A - Contact SDP for T38 reinvite did not match - expected 'c=IN IP4 127.0.0.1' but got [$1]" />
+ </action>
+ </nop>
+
+ <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:[field0]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Testsuite-Track-Phone-A: 2
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901700 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ m=image 10972 udptl t38
+ a=sendrecv
+ a=T38FaxVersion:0
+ a=T38MaxBitRate:9600
+ a=T38FaxMaxBuffer:1024
+ a=T38FaxMaxDatagram:400
+ a=T38FaxRateManagement:transferredTCF
+ a=T38FaxUdpEC:t38UDPRedundancy
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+
+ <!-- Reinvite received when phone B hangs up - media flows between phone A and Asterisk -->
+ <recv request="INVITE"/>
+
+ <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:[field0]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Testsuite-Track-Phone-A: 3
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901700 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ m=image 10972 udptl t38
+ a=sendrecv
+ a=T38FaxVersion:0
+ a=T38MaxBitRate:9600
+ a=T38FaxUdpEC:t38UDPRedundancy
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+
+ <recv request="BYE"/>
+
+ <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:[field0]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Testsuite-Track-Phone-A: 5
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+</scenario>
+
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_A.xml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_A.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_B.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_B.xml?view=auto&rev=4213
==============================================================================
--- asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_B.xml (added)
+++ asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_B.xml Mon Sep 23 12:02:32 2013
@@ -1,0 +1,224 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B Hold with Media Restrictions">
+ <Global variables="remote_tag"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Allow-Events: talk,hold,conference
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <pause milliseconds="200"/>
+
+ <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:[field1]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Testsuite-Track-Phone-B-Media-Restrict: 1
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendrecv
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <!-- RECV ACK -->
+ <recv request="ACK"/>
+
+ <recv request="INVITE">
+ <action>
+ <ereg regexp="c=IN IP4 127.0.0.2" search_in="body" check_it="true" assign_to="1" />
+ <log message="Side B - Contact SDP for directmedia reinvite matches expectations: [$1]." />
+ <strcmp assign_to="result" variable="1" value="c=IN IP4 127.0.0.2" />
+ <test assign_to="emptyinv1" variable="result" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="emptyinv1">
+ <action>
+ <error message="Side B - Contact SDP for directmedia reinvite did not match - expected 'c=IN IP4 127.0.0.2' but got [$1]" />
+ </action>
+ </nop>
+
+ <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:[field1]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Testsuite-Track-Phone-B-Media-Restrict: 2
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendrecv
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="1500"/>
+
+ <!-- Reinvite to set up T38 Fax session -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>
+ CSeq: [cseq] INVITE
+ [last_Call-ID:]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901700 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ m=image 30002 udptl t38
+ a=sendrecv
+ a=T38FaxVersion:0
+ a=T38MaxBitRate:9600
+ a=T38FaxMaxBuffer:1024
+ a=T38FaxMaxDatagram:400
+ a=T38FaxRateManagement:transferredTCF
+ a=T38FaxUdpEC:t38UDPRedundancy
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="200">
+ <action>
+ <ereg regexp="c=IN IP4 127.0.0.1" search_in="body" check_it="true" assign_to="1" />
+ <log message="Side B - Contact SDP for T38 reinvite 200 OK matches expectations: [$1]." />
+ <strcmp assign_to="result" variable="1" value="c=IN IP4 127.0.0.1" />
+ <test assign_to="empty200" variable="result" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="empty200">
+ <action>
+ <error message="Side B - Contact SDP for T38 reinvite 200 OK did not match - expected 'c=IN IP4 127.0.0.1' but got [$1]" />
+ </action>
+ </nop>
+
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ [last_Call-ID:]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="1500"/>
+
+ <send>
+ <![CDATA[
+ BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>[$remote_tag]
+ CSeq: [cseq] BYE
+ [last_Call-ID:]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+
+</scenario>
+
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_B.xml
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Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_B.xml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/endpoint_B.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/inject_bridge.csv
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/inject_bridge.csv?view=auto&rev=4213
==============================================================================
--- asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/inject_bridge.csv (added)
+++ asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/inject_bridge.csv Mon Sep 23 12:02:32 2013
@@ -1,0 +1,3 @@
+SEQUENTIAL
+endpoint_A;endpoint_B;basicdial
+
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/inject_bridge.csv
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Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/sipp/inject_bridge.csv
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svn:keywords = Author Date Id Revision
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/fax/directmedia_reinvite_t38/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/fax/directmedia_reinvite_t38/test-config.yaml?view=auto&rev=4213
==============================================================================
--- asterisk/trunk/tests/fax/directmedia_reinvite_t38/test-config.yaml (added)
+++ asterisk/trunk/tests/fax/directmedia_reinvite_t38/test-config.yaml Mon Sep 23 12:02:32 2013
@@ -1,0 +1,30 @@
+testinfo:
+ summary: 'Test SIP reinvite from directmedia to T.38 FAX'
+ description: |
+ Two devices are in a normal Audio call using directmedia when one does
+ a reinvite to start a T.38 Fax session. This tests that the Fax
+ session reinvite is issued correctly and includes the necessary SDP
+ contact information for Asterisk to be in the path of the fax media.
+
+test-modules:
+ add-test-to-search-path: 'True'
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ fail-on-any: False
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'endpoint_A.xml', '-i': '127.0.0.2', '-p': '5060', '-inf': 'inject_bridge.csv'} }
+ - { 'key-args': {'scenario': 'endpoint_B.xml', '-i': '127.0.0.3', '-p': '5060', '-inf': 'inject_bridge.csv'} }
+
+properties:
+ minversion: '1.8.0.0'
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ tags:
+ - SIP
+ - fax
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/test-config.yaml
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Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/test-config.yaml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/fax/directmedia_reinvite_t38/test-config.yaml
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svn:mime-type = text/plain
Modified: asterisk/trunk/tests/fax/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/fax/tests.yaml?view=diff&rev=4213&r1=4212&r2=4213
==============================================================================
--- asterisk/trunk/tests/fax/tests.yaml (original)
+++ asterisk/trunk/tests/fax/tests.yaml Mon Sep 23 12:02:32 2013
@@ -1,6 +1,7 @@
# Enter tests here in the order they should be considered for execution:
tests:
- test: 'local_channel_t38_queryoption'
+ - test: 'directmedia_reinvite_t38'
- test: 'gateway_g711_t38'
- test: 'gateway_t38_g711'
- test: 'gateway_no_t38'
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