[asterisk-bugs] [JIRA] (ASTERISK-22217) TestSuite sip_hold test fails in SIPp scenarios on unexpected SIP INVITE requests
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Tue Jul 30 13:49:03 CDT 2013
Matt Jordan created ASTERISK-22217:
--------------------------------------
Summary: TestSuite sip_hold test fails in SIPp scenarios on unexpected SIP INVITE requests
Key: ASTERISK-22217
URL: https://issues.asterisk.org/jira/browse/ASTERISK-22217
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: Channels/chan_sip/General, Tests/testsuite
Reporter: Matt Jordan
It appears as if some re-INVITE messages are occurring that weren't before. Yay.
{noformat}
Running ['tests/channels/SIP/sip_hold/run-test'] ...
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'...
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: Done.
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:14:986 1375204274.986483: Aborting call on unexpected message for Call-Id '37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060': while pausing (index 6), received 'INVITE sip:phone_B at 127.0.0.3:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK278ebec0
Max-Forwards: 70
From: "phone_A" <sip:phone_A at 127.0.0.1>;tag=as50c16487
To: <sip:127.0.0.3>;tag=1
Contact: <sip:phone_A at 127.0.0.1:5060>
Call-ID: 37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-trunk-r395686M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 267
v=0
o=root 1387452781 1387452782 IN IP4 127.0.0.2
s=Asterisk PBX SVN-trunk-r395686M
c=IN IP4 127.0.0.2
t=0 0
m=audio 2226 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
'.
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_B_media_restrict.xml Failed [1]
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_B_media_restrict.xml Failed
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'...
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: Done.
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:14:988 1375204274.988396: Aborting call on unexpected message for Call-Id '1-7543 at 127.0.0.2': while expecting 'BYE' (index 6), received 'INVITE sip:phone_A at 127.0.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0c48a3e2
Max-Forwards: 70
From: <sip:basicdial at 127.0.0.1:5060;user=phone>;tag=as004df771
To: phone_A <sip:phone_A at 127.0.0.2:5060>;tag=1
Contact: <sip:basicdial at 127.0.0.1:5060>
Call-ID: 1-7543 at 127.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r395686M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 1242229468 1242229469 IN IP4 127.0.0.1
s=Asterisk PBX SVN-trunk-r395686M
c=IN IP4 127.0.0.1
t=0 0
m=audio 10848 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
'.
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_A.xml Failed [1]
[Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_A.xml Failed
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'...
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: Done.
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:15:193 1375204275.193873: Aborting call on unexpected message for Call-Id '1-7548 at 127.0.0.2': while expecting '180' (index 2), received 'SIP/2.0 603 Declined
Via: SIP/2.0/UDP 127.0.0.2:5060;branch=z9hG4bK-7548-1-0;received=127.0.0.2
From: phone_A <sip:phone_A at 127.0.0.2:5060>;tag=1
To: <sip:basicdial at 127.0.0.1:5060;user=phone>;tag=as4a10f805
Call-ID: 1-7548 at 127.0.0.2
CSeq: 1 INVITE
Server: Asterisk PBX SVN-trunk-r395686M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
'.
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_A.xml Failed [1]
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_A.xml Failed
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'... Done.
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:15:688 1375204275.688405: Aborting call on unexpected message for Call-Id '37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060': while pausing (index 6), received 'INVITE sip:phone_B at 127.0.0.3:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7afe8fbf
Max-Forwards: 70
From: "phone_A" <sip:phone_A at 127.0.0.1>;tag=as50c16487
To: <sip:127.0.0.3>;tag=1
Contact: <sip:phone_A at 127.0.0.1:5060>
Call-ID: 37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX SVN-trunk-r395686M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 267
v=0
o=root 1387452781 1387452783 IN IP4 127.0.0.2
s=Asterisk PBX SVN-trunk-r395686M
c=IN IP4 127.0.0.2
t=0 0
m=audio 2226 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
'.
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_B_IP_restrict.xml Failed [1]
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_B_IP_restrict.xml Failed
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'... Done.
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:15:803 1375204275.803695: Aborting call on unexpected message for Call-Id '1-8043 at 127.0.0.2': while expecting '100' (index 1), received 'SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 127.0.0.2:5060;branch=z9hG4bK-8043-1-0;received=127.0.0.2
From: phone_A <sip:phone_A at 127.0.0.2:5060>;tag=1
To: <sip:basicdial at 127.0.0.1:5060;user=phone>;tag=as44182b0f
Call-ID: 1-8043 at 127.0.0.2
CSeq: 1 INVITE
Server: Asterisk PBX SVN-trunk-r395686M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
'.
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_A.xml Failed [1]
[Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_A.xml Failed
[Jul 30 13:11:19] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'... Done.
[Jul 30 13:11:19] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:19:599 1375204279.599578: Aborting call on unexpected message for Call-Id '37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060': while pausing (index 12), received 'INVITE sip:phone_B at 127.0.0.3:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66e8c55e
Max-Forwards: 70
From: phone_A <sip:phone_B at 127.0.0.1>;tag=as50c16487
To: phone_B <sip:phone_B at 127.0.0.3:5060>;tag=1
Contact: <sip:phone_A at 127.0.0.1:5060>
Call-ID: 37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060
CSeq: 105 INVITE
User-Agent: Asterisk PBX SVN-trunk-r395686M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 1387452781 1387452785 IN IP4 127.0.0.1
s=Asterisk PBX SVN-trunk-r395686M
c=IN IP4 127.0.0.1
t=0 0
m=audio 12068 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=recvonly
'.
[Jul 30 13:11:19] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_B_IP_media_restrict.xml Failed [1]
[Jul 30 13:11:19] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_B_IP_media_restrict.xml Failed
[Jul 30 13:11:20] ERROR[5035]: __main__:116 main: Failed to receive 6 MOH start events (received 0)
[Jul 30 13:11:20] ERROR[5035]: __main__:119 main: Failed to receive 6 MOH stop events (received 0)
[Jul 30 13:11:20] ERROR[5035]: __main__:122 main: Failed to receive 6 user test events (received 4)
{noformat}
--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira
More information about the asterisk-bugs
mailing list