initial commit

pjproject-2.11.1/tests/pjsua/scripts-sendto/999_asterisk_err.py

@@ -0,0 +1,45 @@
+# $Id$
+import inc_sip as sip
+import inc_sdp as sdp
+
+# http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-June/003426.html:
+#
+# Report in pjsip mailing list on 27/6/2008 that this message will
+# cause pjsip to respond with 500 and then second request will cause
+# segfault.
+complete_msg = \
+"""INVITE sip:5001@192.168.1.200:5060;transport=UDP SIP/2.0
+Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK74a60ee5;rport
+From: \"A user\" <sip:66660000@192.168.1.11>;tag=as2858a32c
+To: <sip:5001@192.168.1.200:5060;transport=UDP>
+Contact: <sip:66660000@192.168.1.11>
+Call-ID: 0bc7612c665e875a4a46411442b930a6@192.168.1.11
+CSeq: 102 INVITE
+User-Agent: Asterisk PBX
+Max-Forwards: 70
+Date: Fri, 27 Jun 2008 08:46:47 GMT
+Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
+Supported: replaces
+Content-Type: application/sdp
+Content-Length: 285
+
+v=0
+o=root 4236 4236 IN IP4 192.168.1.11
+s=session
+c=IN IP4 192.168.1.11
+t=0 0
+m=audio 14390 RTP/AVP 0 3 8 101
+a=rtpmap:0 PCMU/8000
+a=rtpmap:3 GSM/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:101 telephone-event/8000
+a=fmtp:101 0-16
+a=silenceSupp:off - - - -
+a=ptime:20
+a=sendrecv
+"""
+
+
+sendto_cfg = sip.SendtoCfg( "Asterisk 500", "--null-audio --auto-answer 200", 
+			    "", 200, complete_msg=complete_msg)
+