rendered paste body*CLI> sip debug peer 2302
SIP Debugging Enabled for IP: 192.168.100.128:5060
*CLI> 11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.168.100.128 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK38fd206f
From: "asterisk" <sip:asterisk@192.168.100.200>;tag=as38a0e281
To: <sip:192.168.100.128>
Contact: <sip:asterisk@192.168.100.200>
Call-ID: 1a98ba375a68da2464bfed5c6023fee9@192.168.100.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 29 Sep 2004 01:27:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 192.168.100.128:5060
Sip read:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK38fd206f
From: "asterisk" <sip:asterisk@192.168.100.200>;tag=as38a0e281
To: <sip:192.168.100.128>;tag=792515087
Contact: <sip:2302@192.168.2.7:5060>
Call-ID: 1a98ba375a68da2464bfed5c6023fee9@192.168.100.200
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY
CSeq: 102 OPTIONS
Server: X-Lite release 1103a
Content-Length: 0
10 headers, 0 lines
Destroying call '1a98ba375a68da2464bfed5c6023fee9@192.168.100.200'
Sip read:
INVITE sip:750@192.168.100.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bKB8BE83275ADD46FFA68A90578F1BF208
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>
Contact: <sip:2302@192.168.2.7:5060>
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7354 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103a
Content-Length: 293
v=0
o=2302 125444890 125445461 IN IP4 192.168.2.7
s=X-Lite
c=IN IP4 192.168.2.7
t=0 0
m=audio 8000 RTP/AVP 3 0 8 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 13 lines
Using latest request as basis request
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.7:5060;branch=z9hG4bKB8BE83275ADD46FFA68A90578F1BF208;received=192.168.100.128;rport=5060
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>;tag=as27059f53
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7354 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:750@192.168.100.200>
Proxy-Authenticate: Digest realm="asterisk", nonce="2bd187ac"
Content-Length: 0
to 192.168.100.128:5060
Scheduling destruction of call 'E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7' in 15000 ms
Found user '2302'
Sip read:
ACK sip:750@192.168.100.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bKB8BE83275ADD46FFA68A90578F1BF208
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>;tag=as27059f53
Contact: <sip:2302@192.168.2.7:5060>
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7354 ACK
Max-Forwards: 70
Content-Length: 0
9 headers, 0 lines
Sip read:
INVITE sip:750@192.168.100.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bK6118392B6CC044E58DE245F5E1E88C2D
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>
Contact: <sip:2302@192.168.2.7:5060>
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7355 INVITE
Proxy-Authorization: Digest username="2302",realm="asterisk",nonce="2bd187ac",response="9e88a2655de56a96d69e857db1c83712",uri="sip:750@192.168.100.200"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103a
Content-Length: 293
v=0
o=2302 125444890 125445461 IN IP4 192.168.2.7
s=X-Lite
c=IN IP4 192.168.2.7
t=0 0
m=audio 8000 RTP/AVP 3 0 8 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
12 headers, 13 lines
Using latest request as basis request
Sending to 192.168.2.7 : 5060 (NAT)
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.7:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x6(GSM|ULAW), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0x6(GSM|ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found user '2302'
Looking for 750 in from-sip
list_route: hop: <sip:2302@192.168.2.7:5060>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.7:5060;branch=z9hG4bK6118392B6CC044E58DE245F5E1E88C2D;received=192.168.100.128;rport=5060
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>;tag=as7fe173ae
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7355 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:750@192.168.100.200>
Content-Length: 0
to 192.168.100.128:5060
We're at 192.168.100.200 port 14388
Answering with preferred capability 0x2(GSM)
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.7:5060;branch=z9hG4bK6118392B6CC044E58DE245F5E1E88C2D;received=192.168.100.128;rport=5060
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>;tag=as7fe173ae
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7355 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:750@192.168.100.200>
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 27383 27383 IN IP4 192.168.100.200
s=session
c=IN IP4 192.168.100.200
t=0 0
m=audio 14388 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 192.168.100.128:5060
Sip read:
ACK sip:750@192.168.100.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bK527A00098AE64F0AA0980DB27C2535E7
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>;tag=as7fe173ae
Contact: <sip:2302@192.168.2.7:5060>
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7355 ACK
Max-Forwards: 70
Content-Length: 0
9 headers, 0 lines
Sep 28 21:27:56 NOTICE[213005]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received
Sep 28 21:28:01 NOTICE[213005]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received
Sip read:
BYE sip:750@192.168.100.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bK6A6F58B7ECFF47EEBD3B7D5953D6F22C
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>;tag=as7fe173ae
Contact: <sip:2302@192.168.2.7:5060>
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7356 BYE
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0
10 headers, 0 lines
Sending to 192.168.2.7 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.7:5060;branch=z9hG4bK6A6F58B7ECFF47EEBD3B7D5953D6F22C;received=192.168.100.128;rport=5060
From: 2302 <sip:2302@192.168.100.200>;tag=2229194259
To: <sip:750@192.168.100.200>;tag=as7fe173ae
Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7
CSeq: 7356 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:750@192.168.100.200>
Content-Length: 0
to 192.168.100.128:5060
Sip read:
0 headers, 0 lines
Destroying call 'E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7'