Merge kamailio modules into sip-router master branch
[sip-router] / test / invite3-callid.sip
1 INVITE sip:p2@iptel.org SIP/2.0
2 Via: SIP/2.0/UDP 193.175.133.193
3 From: "GMD FOKUS iptlab" <sip:jiri@iptel.org>;tag=b96b0300ed30f1286-2f5d
4 To: <sip:p2@iptel.org>
5  Call-ID: b96b0300-88d30f-66da-63aa@195.37.78.190
6 CSeq: 101 INVITE
7 Expires: 180
8 User-Agent: Cisco-SIP-IP-Phone/2
9 Accept: application/sdp
10 Contact: sip:jiri@195.37.78.190:5060
11 Content-Type: application/sdp
12 Content-Length: 225
13
14
15 v=0
16 o=CiscoSystemsSIP-IPPhone-UserAgent 14474 8233 IN IP4 195.37.78.190
17 s=SIP Call
18 c=IN IP4 195.37.78.190
19 t=0 0
20 m=audio 18456 RTP/AVP 0 8 18 101
21 a=rtpmap:0 pcmu/8000
22 a=rtpmap:101 telephone-event/8000
23 a=fmtp:101 0-11