[asterisk-commits] file: testsuite/asterisk/trunk r5008 - in /asterisk/trunk/tests/channels/pjsi...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 1 07:49:19 CDT 2014
Author: file
Date: Thu May 1 07:49:13 2014
New Revision: 5008
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5008
Log:
Add test for ensuring that receiving the same hold SDP multiple times does not cause us to unhold.
ASTERISK-23558 #close
ASTERISK-23558 #comment Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/3473/
Added:
asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_duplicate_hold.xml (with props)
asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_hold_update.xml (with props)
Modified:
asterisk/trunk/tests/channels/pjsip/hold/run-test
Modified: asterisk/trunk/tests/channels/pjsip/hold/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/hold/run-test?view=diff&rev=5008&r1=5007&r2=5008
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/hold/run-test (original)
+++ asterisk/trunk/tests/channels/pjsip/hold/run-test Thu May 1 07:49:13 2014
@@ -39,11 +39,23 @@
'-inf': INJECT_FILE},
{'scenario': 'phone_A.xml',
'-i': '127.0.0.2', '-p': '5060',
+ '-inf': INJECT_FILE},
+ {'scenario': 'phone_A.xml',
+ '-i': '127.0.0.2', '-p': '5060',
+ '-inf': INJECT_FILE},
+ {'scenario': 'phone_A.xml',
+ '-i': '127.0.0.2', '-p': '5060',
'-inf': INJECT_FILE}]
self.sipp_scn_phone_b = [{'scenario': 'phone_B_media_restrict.xml',
'-i': '127.0.0.3', '-p': '5060',
'-inf': INJECT_FILE},
{'scenario': 'phone_B_unhold_sans_sdp.xml',
+ '-i': '127.0.0.3', '-p': '5060',
+ '-inf': INJECT_FILE},
+ {'scenario': 'phone_B_duplicate_hold.xml',
+ '-i': '127.0.0.3', '-p': '5060',
+ '-inf': INJECT_FILE},
+ {'scenario': 'phone_B_hold_update.xml',
'-i': '127.0.0.3', '-p': '5060',
'-inf': INJECT_FILE},
{'scenario': 'phone_B_IP_restrict.xml',
Added: asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_duplicate_hold.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_duplicate_hold.xml?view=auto&rev=5008
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_duplicate_hold.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_duplicate_hold.xml Thu May 1 07:49:13 2014
@@ -1,0 +1,267 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B Hold">
+ <Global variables="global_call_id"/>
+ <Global variables="prime_tag"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ <ereg regexp="tag=.*"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="prime_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
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 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"/>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="2000"/>
+
+ <!-- Modify RTP session to be send only -->
+ <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]>;[$prime_tag]
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_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=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendonly
+ m=audio 2226 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="200" />
+
+ <pause milliseconds="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: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="2000"/>
+
+ <!-- Modify RTP session to be send only -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>;[$prime_tag]
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendonly
+ m=audio 2226 RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="200" />
+
+ <pause milliseconds="200"/>
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_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, then send the un-hold as reinvite without SDP -->
+ <pause milliseconds="2000"/>
+
+ <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]>;[$prime_tag]
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_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-Length: 0
+ ]]>
+ </send>
+
+ <recv response="100" 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: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="2000"/>
+
+ <send>
+ <![CDATA[
+ BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [$global_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/channels/pjsip/hold/sipp/phone_B_duplicate_hold.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_duplicate_hold.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_duplicate_hold.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_hold_update.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_hold_update.xml?view=auto&rev=5008
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_hold_update.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/hold/sipp/phone_B_hold_update.xml Thu May 1 07:49:13 2014
@@ -1,0 +1,267 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B Hold">
+ <Global variables="global_call_id"/>
+ <Global variables="prime_tag"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ <ereg regexp="tag=.*"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="prime_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
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 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"/>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="2000"/>
+
+ <!-- Modify RTP session to be send only -->
+ <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]>;[$prime_tag]
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_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=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendonly
+ m=audio 2226 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="200" />
+
+ <pause milliseconds="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: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="2000"/>
+
+ <!-- Modify RTP session to be send only -->
+ <send retrans="500">
+ <![CDATA[
+ UPDATE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>;[$prime_tag]
+ CSeq: [cseq] UPDATE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendonly
+ m=audio 2226 RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="200" />
+
+ <pause milliseconds="200"/>
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_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, then send the un-hold as reinvite without SDP -->
+ <pause milliseconds="2000"/>
+
+ <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]>;[$prime_tag]
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_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-Length: 0
+ ]]>
+ </send>
+
+ <recv response="100" 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: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="2000"/>
+
+ <send>
+ <![CDATA[
+ BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [$global_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>
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