blob: 580b46de41e477880e9bf80400a5557950aca80b [file] [log] [blame]
Benny Prijonoab8dba92008-06-27 21:59:15 +00001# $Id$
2import inc_sip as sip
3import inc_sdp as sdp
4
5# http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-June/003426.html:
6#
7# Report in pjsip mailing list on 27/6/2008 that this message will
8# cause pjsip to respond with 500 and then second request will cause
9# segfault.
10complete_msg = \
11"""INVITE sip:5001@192.168.1.200:5060;transport=UDP SIP/2.0
12Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK74a60ee5;rport
13From: \"A user\" <sip:66660000@192.168.1.11>;tag=as2858a32c
14To: <sip:5001@192.168.1.200:5060;transport=UDP>
15Contact: <sip:66660000@192.168.1.11>
16Call-ID: 0bc7612c665e875a4a46411442b930a6@192.168.1.11
17CSeq: 102 INVITE
18User-Agent: Asterisk PBX
19Max-Forwards: 70
20Date: Fri, 27 Jun 2008 08:46:47 GMT
21Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
22Supported: replaces
23Content-Type: application/sdp
24Content-Length: 285
25
26v=0
27o=root 4236 4236 IN IP4 192.168.1.11
28s=session
29c=IN IP4 192.168.1.11
30t=0 0
31m=audio 14390 RTP/AVP 0 3 8 101
32a=rtpmap:0 PCMU/8000
33a=rtpmap:3 GSM/8000
34a=rtpmap:8 PCMA/8000
35a=rtpmap:101 telephone-event/8000
36a=fmtp:101 0-16
37a=silenceSupp:off - - - -
38a=ptime:20
39a=sendrecv
40"""
41
42
43sendto_cfg = sip.SendtoCfg( "Asterisk 500", "--null-audio --auto-answer 200",
44 "", 200, complete_msg=complete_msg)
45