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Viewing changes to sflphone-common/libs/pjproject/tests/pjsua/scripts-sendto/999_asterisk_err.py

  • Committer: Bazaar Package Importer
  • Author(s): Francois Marier
  • Date: 2010-06-03 15:59:46 UTC
  • Revision ID: james.westby@ubuntu.com-20100603155946-ybe8d8o8zx8lp0m8
Tags: upstream-0.9.8.3
Import upstream version 0.9.8.3

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sflphone-common/libs/pjproject/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)
 
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