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Viewing changes to daemon/libs/pjproject-2.0.1/tests/pjsua/scripts-sendto/999_asterisk_err.py

  • Committer: Package Import Robot
  • Author(s): Mark Purcell
  • Date: 2014-01-28 18:23:36 UTC
  • mfrom: (4.3.4 sid)
  • Revision ID: package-import@ubuntu.com-20140128182336-jrsv0k9u6cawc068
Tags: 1.3.0-1
http://bugs.debian.org/735846
http://bugs.debian.org/727164
http://bugs.debian.org/718178
http://bugs.debian.org/736583
http://bugs.debian.org/722040
http://bugs.debian.org/736746
* New upstream release 
  - Fixes "New Upstream Release" (Closes: #735846)
  - Fixes "Ringtone does not stop" (Closes: #727164)
  - Fixes "[sflphone-kde] crash on startup" (Closes: #718178)
  - Fixes "sflphone GUI crashes when call is hung up" (Closes: #736583)
* Build-Depends: ensure GnuTLS 2.6
  - libucommon-dev (>= 6.0.7-1.1), libccrtp-dev (>= 2.0.6-3)
  - Fixes "FTBFS Build-Depends libgnutls{26,28}-dev" (Closes: #722040)
* Fix "boost 1.49 is going away" unversioned Build-Depends: (Closes: #736746)
* Add Build-Depends: libsndfile-dev, nepomuk-core-dev

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daemon/libs/pjproject-2.0.1/tests/pjsua/scripts-sendto/999_asterisk_err.py
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# $Id: 999_asterisk_err.py 2081 2008-06-27 21:59:15Z bennylp $
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import inc_sip as sip
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import inc_sdp as sdp
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# http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-June/003426.html:
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#
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# Report in pjsip mailing list on 27/6/2008 that this message will
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# cause pjsip to respond with 500 and then second request will cause
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# segfault.
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complete_msg = \
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"""INVITE sip:5001@192.168.1.200:5060;transport=UDP SIP/2.0
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Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK74a60ee5;rport
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From: \"A user\" <sip:66660000@192.168.1.11>;tag=as2858a32c
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To: <sip:5001@192.168.1.200:5060;transport=UDP>
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Contact: <sip:66660000@192.168.1.11>
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Call-ID: 0bc7612c665e875a4a46411442b930a6@192.168.1.11
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CSeq: 102 INVITE
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User-Agent: Asterisk PBX
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Max-Forwards: 70
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Date: Fri, 27 Jun 2008 08:46:47 GMT
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Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
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Supported: replaces
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Content-Type: application/sdp
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Content-Length: 285
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v=0
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o=root 4236 4236 IN IP4 192.168.1.11
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s=session
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c=IN IP4 192.168.1.11
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t=0 0
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m=audio 14390 RTP/AVP 0 3 8 101
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a=rtpmap:0 PCMU/8000
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a=rtpmap:3 GSM/8000
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a=rtpmap:8 PCMA/8000
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a=rtpmap:101 telephone-event/8000
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a=fmtp:101 0-16
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a=silenceSupp:off - - - -
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a=ptime:20
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a=sendrecv
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"""
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sendto_cfg = sip.SendtoCfg( "Asterisk 500", "--null-audio --auto-answer 200",
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                            "", 200, complete_msg=complete_msg)