Tristan Matthews | 0a329cc | 2013-07-17 13:20:14 -0400 | [diff] [blame] | 1 | # $Id$ |
| 2 | import inc_sip as sip |
| 3 | import 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. |
| 10 | complete_msg = \ |
| 11 | """INVITE sip:5001@192.168.1.200:5060;transport=UDP SIP/2.0 |
| 12 | Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK74a60ee5;rport |
| 13 | From: \"A user\" <sip:66660000@192.168.1.11>;tag=as2858a32c |
| 14 | To: <sip:5001@192.168.1.200:5060;transport=UDP> |
| 15 | Contact: <sip:66660000@192.168.1.11> |
| 16 | Call-ID: 0bc7612c665e875a4a46411442b930a6@192.168.1.11 |
| 17 | CSeq: 102 INVITE |
| 18 | User-Agent: Asterisk PBX |
| 19 | Max-Forwards: 70 |
| 20 | Date: Fri, 27 Jun 2008 08:46:47 GMT |
| 21 | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY |
| 22 | Supported: replaces |
| 23 | Content-Type: application/sdp |
| 24 | Content-Length: 285 |
| 25 | |
| 26 | v=0 |
| 27 | o=root 4236 4236 IN IP4 192.168.1.11 |
| 28 | s=session |
| 29 | c=IN IP4 192.168.1.11 |
| 30 | t=0 0 |
| 31 | m=audio 14390 RTP/AVP 0 3 8 101 |
| 32 | a=rtpmap:0 PCMU/8000 |
| 33 | a=rtpmap:3 GSM/8000 |
| 34 | a=rtpmap:8 PCMA/8000 |
| 35 | a=rtpmap:101 telephone-event/8000 |
| 36 | a=fmtp:101 0-16 |
| 37 | a=silenceSupp:off - - - - |
| 38 | a=ptime:20 |
| 39 | a=sendrecv |
| 40 | """ |
| 41 | |
| 42 | |
| 43 | sendto_cfg = sip.SendtoCfg( "Asterisk 500", "--null-audio --auto-answer 200", |
| 44 | "", 200, complete_msg=complete_msg) |
| 45 | |