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<DIV><FONT face=Arial size=2>Hi Clod,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The test bed and configuration is very simple. The
call was completed but the CDR function did not work. I do not know what I
missed.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><STRONG>test bed:</STRONG></FONT></DIV>
<DIV><FONT face=Arial size=2>source (sipp at 127.0.0.1:5062) -> asterisk
(127.0.0.1:5061) -> destination (sipp at 127.0.0.1:5060)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>sip.conf</FONT></DIV>
<DIV><FONT face=Arial size=2>[general]</FONT></DIV>
<DIV><FONT face=Arial size=2>context=GeneralProxy</FONT></DIV>
<DIV><FONT face=Arial size=2>allowguest=yes</FONT></DIV>
<DIV><FONT face=Arial size=2>bindport=5061</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><STRONG>extensions.conf</STRONG></FONT></DIV>
<DIV><FONT face=Arial size=2>[GeneralProxy]<BR>exten =>
_XXXX.,1,NoOp(GeneralProxy)<BR>exten => _XXXX.,n,Dial(<A
href="mailto:SIP/1234567890@127.0.0.1,14,oL(14400000">SIP/1234567890@127.0.0.1,14,oL(14400000</A>]))<BR>exten
=> _XXXX.,n,Hangup<BR>exten => h,1,NoOp()<BR>exten =>
h,n,Set(foo=${CDR(src)})</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><STRONG>The CLI log</STRONG></FONT></DIV>
<DIV><FONT face=Arial size=2><--- SIP read from 127.0.0.1:5062
---><BR>INVITE sip:1234567890@127.0.0.1:5061 SIP/2.0<BR>Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK-27105-1-0<BR>From: sipp
<sip:sipp@127.0.0.1:5062>;tag=27105SIPpTag001<BR>To: sut
<sip:1234567890@127.0.0.1:5061><BR>Call-ID: <A
href="mailto:1-27105@127.0.0.1">1-27105@127.0.0.1</A><BR>CSeq: 1
INVITE<BR>Contact: sip:sipp@127.0.0.1:5062<BR>Max-Forwards: 70<BR>Subject:
Performance Test<BR>Content-Type: application/sdp<BR>Content-Length:
129</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=user1 53655765 2353687637 IN IP4
127.0.0.1<BR>s=-<BR>c=IN IP4 127.0.0.1<BR>t=0 0<BR>m=audio 6001 RTP/AVP
0<BR>a=rtpmap:0 PCMU/8000</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2><-------------><BR>--- (11 headers 7 lines)
---<BR>Sending to 127.0.0.1 : 5062 (NAT)<BR>Using INVITE request as basis
request - <A href="mailto:1-27105@127.0.0.1">1-27105@127.0.0.1</A><BR>Found no
matching peer or user for '127.0.0.1:5062'<BR>Found RTP audio format 0<BR>Peer
audio RTP is at port 127.0.0.1:6001<BR>Found description format PCMU for ID
0<BR>Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)<BR>Non-codec capabilities
(dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0
(nothing)<BR>Peer audio RTP is at port 127.0.0.1:6001<BR>Looking for 1234567890
in GeneralProxy (domain 127.0.0.1)<BR>list_route: hop:
<sip:sipp@127.0.0.1:5062></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2><--- Transmitting (NAT) to 127.0.0.1:5062
---><BR>SIP/2.0 100 Trying<BR>Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK-27105-1-0;received=127.0.0.1<BR>From: sipp
<sip:sipp@127.0.0.1:5062>;tag=27105SIPpTag001<BR>To: sut
<sip:1234567890@127.0.0.1:5061><BR>Call-ID: <A
href="mailto:1-27105@127.0.0.1">1-27105@127.0.0.1</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Supported: replaces<BR>Contact:
<sip:1234567890@127.0.0.1:5061><BR>Content-Length: 0</FONT></DIV>
<DIV> </DIV><FONT face=Arial size=2>
<DIV><BR><------------><BR> -- Executing
[1234567890@GeneralProxy:1] NoOp("SIP/5062-083d8d78", "GeneralProxy") in new
stack<BR> -- Executing [1234567890@GeneralProxy:2]
Dial("SIP/5062-083d8d78", "<A
href="mailto:SIP/1234567890@127.0.0.1|14|oL(14400000">SIP/1234567890@127.0.0.1|14|oL(14400000</A>])")
in new stack<BR> -- Setting call duration limit to 14400
seconds.<BR>Audio is at 127.0.0.1 port 14012<BR>Adding codec 0x4 (ulaw) to
SDP<BR>Adding codec 0x2 (gsm) to SDP<BR>Adding codec 0x8 (alaw) to SDP<BR>Adding
non-codec 0x1 (telephone-event) to SDP<BR>Reliably Transmitting (NAT) to
127.0.0.1:5060:<BR>INVITE sip:1234567890@127.0.0.1 SIP/2.0<BR>Via: SIP/2.0/UDP
127.0.0.1:5061;branch=z9hG4bK7c6c880d;rport<BR>From: "sipp"
<sip:sipp@127.0.0.1:5061>;tag=as31d2fe73<BR>To:
<sip:1234567890@127.0.0.1><BR>Contact:
<sip:sipp@127.0.0.1:5061><BR>Call-ID: <A
href="mailto:43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1">43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Thu, 05 Apr
2007 06:17:52 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY<BR>Supported: replaces<BR>Content-Type:
application/sdp<BR>Content-Length: 281</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 27084 27084 IN IP4 127.0.0.1<BR>s=session<BR>c=IN IP4
127.0.0.1<BR>t=0 0<BR>m=audio 14012 RTP/AVP 0 3 8 101<BR>a=rtpmap:0
PCMU/8000<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - -
-<BR>a=ptime:20<BR>a=sendrecv</DIV>
<DIV> </DIV>
<DIV>---<BR> -- Called <A
href="mailto:1234567890@127.0.0.1">1234567890@127.0.0.1</A></DIV>
<DIV> </DIV>
<DIV><--- SIP read from 127.0.0.1:5060 ---><BR>SIP/2.0 180 Ringing<BR>Via:
SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK7c6c880d;rport<BR>From: "sipp"
<sip:sipp@127.0.0.1:5061>;tag=as31d2fe73<BR>To:
<sip:1234567890@127.0.0.1>;tag=11675SIPpTag018<BR>Call-ID: <A
href="mailto:43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1">43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1</A><BR>CSeq:
102 INVITE<BR>Contact:
<sip:127.0.0.1:5060;transport=UDP><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR><-------------><BR>--- (8 headers 0 lines)
---<BR> -- SIP/127.0.0.1-083ddd70 is ringing</DIV>
<DIV> </DIV>
<DIV><--- SIP read from 127.0.0.1:5060 ---><BR>SIP/2.0 200 OK<BR>Via:
SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK7c6c880d;rport<BR>From: "sipp"
<sip:sipp@127.0.0.1:5061>;tag=as31d2fe73<BR>To:
<sip:1234567890@127.0.0.1>;tag=11675SIPpTag018<BR>Call-ID: <A
href="mailto:43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1">43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1</A><BR>CSeq:
102 INVITE<BR>Contact: <sip:127.0.0.1:5060;transport=UDP><BR>Content-Type:
application/sdp<BR>Content-Length: 129</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=user1 53655765 2353687637 IN IP4 127.0.0.1<BR>s=-<BR>c=IN IP4
127.0.0.1<BR>t=0 0<BR>m=audio 6000 RTP/AVP 0<BR>a=rtpmap:0 PCMU/8000</DIV>
<DIV> </DIV>
<DIV><-------------><BR>--- (9 headers 7 lines) ---<BR>Found RTP audio
format 0<BR>Peer audio RTP is at port 127.0.0.1:6000<BR>Found description format
PCMU for ID 0<BR>Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer -
audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)<BR>Non-codec
capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined
- 0x0 (nothing)<BR>Peer audio RTP is at port 127.0.0.1:6000<BR>list_route: hop:
<sip:127.0.0.1:5060;transport=UDP><BR>set_destination: Parsing
<sip:127.0.0.1:5060;transport=UDP> for address/port to send
to<BR>set_destination: set destination to 127.0.0.1, port 5060<BR>Transmitting
(NAT) to 127.0.0.1:5060:<BR>ACK sip:127.0.0.1:5060;transport=UDP SIP/2.0<BR>Via:
SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK33206146;rport<BR>From: "sipp"
<sip:sipp@127.0.0.1:5061>;tag=as31d2fe73<BR>To:
<sip:1234567890@127.0.0.1>;tag=11675SIPpTag018<BR>Contact:
<sip:sipp@127.0.0.1:5061><BR>Call-ID: <A
href="mailto:43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1">43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1</A><BR>CSeq:
102 ACK<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>---</DIV>
<DIV> </DIV>
<DIV><--- Transmitting (NAT) to 127.0.0.1:5062 ---><BR>SIP/2.0 180
Ringing<BR>Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK-27105-1-0;received=127.0.0.1<BR>From: sipp
<sip:sipp@127.0.0.1:5062>;tag=27105SIPpTag001<BR>To: sut
<sip:1234567890@127.0.0.1:5061>;tag=as58fef656<BR>Call-ID: <A
href="mailto:1-27105@127.0.0.1">1-27105@127.0.0.1</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Supported: replaces<BR>Contact:
<sip:1234567890@127.0.0.1:5061><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR><------------><BR> -- Call on
SIP/127.0.0.1-083ddd70 left from hold<BR> --
SIP/127.0.0.1-083ddd70 answered SIP/5062-083d8d78<BR>Audio is at 127.0.0.1 port
11154<BR>Adding codec 0x4 (ulaw) to SDP</DIV>
<DIV> </DIV>
<DIV><--- Reliably Transmitting (NAT) to 127.0.0.1:5062 ---><BR>SIP/2.0
200 OK<BR>Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK-27105-1-0;received=127.0.0.1<BR>From: sipp
<sip:sipp@127.0.0.1:5062>;tag=27105SIPpTag001<BR>To: sut
<sip:1234567890@127.0.0.1:5061>;tag=as58fef656<BR>Call-ID: <A
href="mailto:1-27105@127.0.0.1">1-27105@127.0.0.1</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Supported: replaces<BR>Contact:
<sip:1234567890@127.0.0.1:5061><BR>Content-Type:
application/sdp<BR>Content-Length: 178</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 27084 27084 IN IP4 127.0.0.1<BR>s=session<BR>c=IN IP4
127.0.0.1<BR>t=0 0<BR>m=audio 11154 RTP/AVP 0<BR>a=rtpmap:0
PCMU/8000<BR>a=silenceSupp:off - - - -<BR>a=ptime:20<BR>a=sendrecv</DIV>
<DIV> </DIV>
<DIV><------------><BR>[Apr 5 14:17:52] WARNING[27109]: cdr.c:482
ast_cdr_merge: CDR start disagreement for
SIP/5062-083d8d78<BR> -- Packet2Packet bridging
SIP/5062-083d8d78 and SIP/127.0.0.1-083ddd70</DIV>
<DIV> </DIV>
<DIV><--- SIP read from 127.0.0.1:5062 ---><BR>ACK
sip:1234567890@127.0.0.1:5061 SIP/2.0<BR>Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK-27105-1-4<BR>From: sipp
<sip:sipp@127.0.0.1:5062>;tag=27105SIPpTag001<BR>To: sut
<sip:1234567890@127.0.0.1:5061>;tag=as58fef656<BR>Call-ID: <A
href="mailto:1-27105@127.0.0.1">1-27105@127.0.0.1</A><BR>CSeq: 1 ACK<BR>Contact:
sip:sipp@127.0.0.1:5062<BR>Max-Forwards: 70<BR>Subject: Performance
Test<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR><-------------><BR>--- (10 headers 0 lines) ---</DIV>
<DIV> </DIV>
<DIV><--- SIP read from 127.0.0.1:5062 ---><BR>BYE
sip:1234567890@127.0.0.1:5061 SIP/2.0<BR>Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK-27105-1-6<BR>From: sipp
<sip:sipp@127.0.0.1:5062>;tag=27105SIPpTag001<BR>To: sut
<sip:1234567890@127.0.0.1:5061>;tag=as58fef656<BR>Call-ID: <A
href="mailto:1-27105@127.0.0.1">1-27105@127.0.0.1</A><BR>CSeq: 2 BYE<BR>Contact:
sip:sipp@127.0.0.1:5062<BR>Max-Forwards: 70<BR>Subject: Performance
Test<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR><-------------><BR>--- (10 headers 0 lines) ---<BR>Sending to
127.0.0.1 : 5062 (NAT)</DIV>
<DIV> </DIV>
<DIV><--- Transmitting (NAT) to 127.0.0.1:5062 ---><BR>SIP/2.0 200
OK<BR>Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK-27105-1-6;received=127.0.0.1<BR>From: sipp
<sip:sipp@127.0.0.1:5062>;tag=27105SIPpTag001<BR>To: sut
<sip:1234567890@127.0.0.1:5061>;tag=as58fef656<BR>Call-ID: <A
href="mailto:1-27105@127.0.0.1">1-27105@127.0.0.1</A><BR>CSeq: 2
BYE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Supported: replaces<BR>Contact:
<sip:1234567890@127.0.0.1:5061><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR><------------><BR>Scheduling destruction of SIP dialog <A
href="mailto:'43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1'">'43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1'</A>
in 32000 ms (Method: INVITE)<BR>set_destination: Parsing
<sip:127.0.0.1:5060;transport=UDP> for address/port to send
to<BR>set_destination: set destination to 127.0.0.1, port 5060<BR>Reliably
Transmitting (NAT) to 127.0.0.1:5060:<BR>BYE sip:127.0.0.1:5060;transport=UDP
SIP/2.0<BR>Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK4e1dac73;rport<BR>From:
"sipp" <sip:sipp@127.0.0.1:5061>;tag=as31d2fe73<BR>To:
<sip:1234567890@127.0.0.1>;tag=11675SIPpTag018<BR>Call-ID: <A
href="mailto:43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1">43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1</A><BR>CSeq:
103 BYE<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>---<BR> == Spawn extension (GeneralProxy, 1234567890, 2) exited
non-zero on 'SIP/5062-083d8d78'<BR> -- Executing
[h@GeneralProxy:1] NoOp("SIP/5062-083d8d78", "") in new
stack<BR> -- Executing [h@GeneralProxy:2]
Set("SIP/5062-083d8d78", <FONT color=#ff0000>"foo="</FONT>) in new stack</DIV>
<DIV> </DIV>
<DIV><--- SIP read from 127.0.0.1:5060 ---><BR>SIP/2.0 200 OK<BR>Via:
SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK4e1dac73;rport<BR>From: "sipp"
<sip:sipp@127.0.0.1:5061>;tag=as31d2fe73<BR>To:
<sip:1234567890@127.0.0.1>;tag=11675SIPpTag018<BR>Call-ID: <A
href="mailto:43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1">43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1</A><BR>CSeq:
103 BYE<BR>Contact: <sip:127.0.0.1:5060;transport=UDP><BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR><-------------><BR>--- (8 headers 0 lines) ---<BR>Really
destroying SIP dialog <A
href="mailto:'43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1'">'43e1d73239f5d00f2abf2d111f732cb0@127.0.0.1'</A>
Method: INVITE<BR>Really destroying SIP dialog <A
href="mailto:'1-27105@127.0.0.1'">'1-27105@127.0.0.1'</A> Method: BYE</DIV>
<DIV> </DIV>
<DIV>Thanks,</DIV>
<DIV> </DIV>
<DIV>Di-Shi Sun.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=cpatry@gmail.com href="mailto:cpatry@gmail.com">Clod Patry</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A title=di-shi@transnexus.com
href="mailto:di-shi@transnexus.com">Di-Shi Sun</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, April 05, 2007 11:11
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [asterisk-dev] cdr of sip
channel</DIV>
<DIV><BR></DIV>no, since i dont know what ur setup/dialplan looks
like.<BR><BR>
<DIV><SPAN class=gmail_quote>On 4/4/07, <B class=gmail_sendername>Di-Shi
Sun</B> <<A
href="mailto:di-shi@transnexus.com">di-shi@transnexus.com</A>> wrote:
</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV bgcolor="#ffffff">
<DIV><FONT face=Arial size=2>Hi Clod,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thank you for your reply. I had tried the CDR
fucntion in my dialplan. But it retuened nothing.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Any idea?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Di-Shi Sun.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2></FONT> </DIV></DIV></BLOCKQUOTE></DIV><BR><BR clear=all><BR>--
<BR>Clod Patry </BLOCKQUOTE></BODY></HTML>