SRVVOIP1*CLI> sip set debug peer 200
SIP Debugging Enabled for IP: 10.200.X.X:8020
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
INVITE sip:00326XXXXXX@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-ca1bf268ec73c165-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@10.200.X.X:8020>
To: <sip:00326XXXXXX@DOMAINE>
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
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: 370
v=0
o=- 12961590236339587 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:50c323
a=ice-pwd:f1e6b24f672647dfb9ae17ad5d151de7
m=audio 8024 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.200.X.X 8024 typ host
a=candidate:1 2 UDP 659134 10.200.X.X 8025 typ host
<------------->
--- (13 headers 13 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.200.X.X : 8020 (NAT)
Using INVITE request as basis request - NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
Found user '200' for '200'
SRVVOIP1*CLI>
<--- Reliably Transmitting (no NAT) to 10.200.X.X:8020 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-ca1bf268ec73c165-1---d8754z-;received=10.200.10.44;rport=8020
From: "bob"<sip:200@nestal.localnet>;tag=0394c0b8
To: <sip:00326XXXXXX@DOMAINE>;tag=as02783670
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
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="1d152016"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
ACK sip:00326XXXXXX@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-ca1bf268ec73c165-1---d8754z-;rport
Max-Forwards: 70
To: <sip:00326XXXXXX@DOMAINE>;tag=as02783670
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
INVITE sip:00326XXXXXX@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-1e090f4007c8f285-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@10.200.X.X:8020>
To: <sip:00326XXXXXX@DOMAINE>
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
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="1d152016",u ri="sip:00326XXXXXX@DOMAINE",response="1825e8189dc eeb1edf2ca0b072fa817b",algorithm=MD5
Content-Length: 370
v=0
o=- 12961590236339587 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:50c323
a=ice-pwd:f1e6b24f672647dfb9ae17ad5d151de7
m=audio 8024 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.200.X.X 8024 typ host
a=candidate:1 2 UDP 659134 10.200.X.X 8025 typ host
<------------->
--- (14 headers 13 lines) ---
Sending to 10.200.X.X : 8020 (no NAT)
Using INVITE request as basis request - NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
Found user '200' for '200'
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/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)
SRVVOIP1*CLI>
<--- Reliably Transmitting (no NAT) to 10.200.X.X:8020 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-1e090f4007c8f285-1---d8754z-;received=10.200.X.X;rport=8020
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
To: <sip:00326XXXXXX@DOMAINE>;tag=as02783670
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
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 'NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
ACK sip:00326XXXXXX@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-1e090f4007c8f285-1---d8754z-;rport
Max-Forwards: 70
To: <sip:00326XXXXXX@DOMAINE>;tag=as02783670
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[SRVVOIP1.localdomain asterisk]#
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