rendered paste body -- Executing [s@macro-dial:10] Dial("Zap/2-1", "SIP/101|20|tr") in new stack
Video is at 192.168.16.180 port 14568
Audio is at 192.168.16.180 port 18260
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.16.184:22478:
INVITE sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.180:5060;branch=z9hG4bK0df3a0dc;rport
From: "Unknown" <sip:Unknown@192.168.16.180>;tag=as0c3ec5a9
To: <sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1>
Contact: <sip:Unknown@192.168.16.180>
Call-ID: 26b52eaf193d06e709d1893d2d447c48@192.168.16.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 07 Jun 2007 11:13:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 374
v=0
o=root 10484 10484 IN IP4 192.168.16.180
s=session
c=IN IP4 192.168.16.180
b=CT:384
t=0 0
m=audio 18260 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 14568 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=sendrecv
---
-- Called 101
Retransmitting #1 (NAT) to 192.168.16.184:22478:
INVITE sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.180:5060;branch=z9hG4bK0df3a0dc;rport
From: "Unknown" <sip:Unknown@192.168.16.180>;tag=as0c3ec5a9
To: <sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1>
Contact: <sip:Unknown@192.168.16.180>
Call-ID: 26b52eaf193d06e709d1893d2d447c48@192.168.16.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 07 Jun 2007 11:13:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 374
v=0
o=root 10484 10484 IN IP4 192.168.16.180
s=session
c=IN IP4 192.168.16.180
b=CT:384
t=0 0
m=audio 18260 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 14568 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=sendrecv
---
astbath*CLI>
<--- SIP read from 192.168.16.184:22478 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.16.180:5060;branch=z9hG4bK0df3a0dc;rport=5060
Contact: <sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1>
To: <sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1>;tag=b34b9757
From: "Unknown"<sip:Unknown@192.168.16.180>;tag=as0c3ec5a9
Call-ID: 26b52eaf193d06e709d1893d2d447c48@192.168.16.180
CSeq: 102 INVITE
User-Agent: eyeBeam release 1011s stamp 41121
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/101-09857148 is ringing
astbath*CLI>
<--- SIP read from 192.168.16.184:22478 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.16.180:5060;branch=z9hG4bK0df3a0dc;rport=5060
Contact: <sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1>
To: <sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1>;tag=b34b9757
From: "Unknown"<sip:Unknown@192.168.16.180>;tag=as0c3ec5a9
Call-ID: 26b52eaf193d06e709d1893d2d447c48@192.168.16.180
CSeq: 102 INVITE
User-Agent: eyeBeam release 1011s stamp 41121
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/101-09857148 is ringing
-- Channel 0/2, span 1 got hangup request
Scheduling destruction of SIP dialog '26b52eaf193d06e709d1893d2d447c48@192.168.16.180' in 6592 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.16.184:22478:
CANCEL sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.180:5060;branch=z9hG4bK0df3a0dc;rport
From: "Unknown" <sip:Unknown@192.168.16.180>;tag=as0c3ec5a9
To: <sip:101@192.168.16.184:22478;rinstance=4da0d470daba5fd1>
Call-ID: 26b52eaf193d06e709d1893d2d447c48@192.168.16.180
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0