SRVVOIP1*CLI> sip set debug peer 200
SIP Debugging Enabled for IP: 10.200.X.X:8024
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8024 --->
INVITE sip:202@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-186b4174337703ac-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@10.200.X.X:8024>
To: <sip:202@DOMAINE>
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 488
v=0
o=- 12961589563258089 1 IN IP4 10.200.X.X
s=CounterPath X-Lite 4.1
c=IN IP4 10.200.X.X
t=0 0
a=ice-ufrag:a323b6
a=ice-pwd:3d8bb9985126dad4d71a6bace385a126
m=audio 8058 RTP/AVP 100 97 105 98 3 101
a=rtpmap:100 SPEEX/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.200.X.X 8058 typ host
a=candidate:1 2 UDP 659134 10.200.X.X 8059 typ host
<------------->
--- (13 headers 17 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.200.X.X : 8024 (NAT)
Using INVITE request as basis request - MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
Found user '200' for '200'
<--- Reliably Transmitting (no NAT) to 10.200.X.X:8024 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-186b4174337703ac-1---d8754z-;received=10.200.10.44;rport=8024
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
To: <sip:202@DOMAINE>;tag=as25f0b610
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61a5b176"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.' in 32000 ms (Method: INVITE)
<--- SIP read from UDP://10.200.X.X:8024 --->
ACK sip:202@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-186b4174337703ac-1---d8754z-;rport
Max-Forwards: 70
To: <sip:202@DOMAINE>;tag=as25f0b610
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8024 --->
INVITE sip:202@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-425621f5fe1674ee-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@10.200.X.X:8024>
To: <sip:202@DOMAINE>
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="200",realm="asterisk",nonce="61a5b176",u ri="sip:202@DOMAINE",response="e0067352b88987ff830 1fee4507c6be6",algorithm=MD5
Content-Length: 488
v=0
o=- 12961589563258089 1 IN IP4 10.200.X.X
s=CounterPath X-Lite 4.1
c=IN IP4 10.200.X.X
t=0 0
a=ice-ufrag:a323b6
a=ice-pwd:3d8bb9985126dad4d71a6bace385a126
m=audio 8058 RTP/AVP 100 97 105 98 3 101
a=rtpmap:100 SPEEX/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.200.X.X 8058 typ host
a=candidate:1 2 UDP 659134 10.200.X.X 8059 typ host
<------------->
--- (14 headers 17 lines) ---
Sending to 10.200.X.X : 8024 (no NAT)
Using INVITE request as basis request - MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
Found user '200' for '200'
Found RTP audio format 100
Found RTP audio format 97
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 3
Found RTP audio format 101
Found audio description format SPEEX for ID 100
Found audio description format SPEEX for ID 97
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x602 (gsm|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
<--- Reliably Transmitting (no NAT) to 10.200.X.X:8024 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-425621f5fe1674ee-1---d8754z-;received=10.200.X.X;rport=8024
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
To: <sip:202@DOMAINE>;tag=as25f0b610
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8024 --->
ACK sip:202@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-425621f5fe1674ee-1---d8754z-;rport
Max-Forwards: 70
To: <sip:202@DOMAINE>;tag=as25f0b610
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'NWJlYWNhOTkwYmEzZjc4MWIwZTJlNDM3NTk1MjIzZTA.' Method: ACK
SRVVOIP1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[SRVVOIP1.localdomain asterisk]#