uneu*CLI> sip debug
SIP Debugging Enabled
uneu*CLI>
Sip read:
INVITE sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bK4ea1f6981249f975
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 33612 INVITE
User-Agent: Grandstream HT486 1.0.5.10
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 315
v=0
o=70 8000 8000 IN IP4 213.54.212.41
s=SIP Call
c=IN IP4 213.54.212.41
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 2 15 4
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:4 G723/8000
a=ptime:40
12 headers, 15 lines
Using latest request as basis request
Sending to 213.54.212.41 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found audio format ULAW
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G726-32
Found description format G728
Found description format G723
Capabilities: us - 1030, them - 1309/0, combined - 1028
Non-codec capabilities: us - 1, them - 0, combined - 0
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bK4ea1f6981249f975
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>;tag=as1f32ff34
Call-ID: [email protected]
CSeq: 33612 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*[email protected]>
Proxy-Authenticate: Digest realm="asterisk", nonce="167fb37c"
Content-Length: 0
to 213.54.212.41:5060
uneu*CLI>
Sip read:
ACK sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bK4ea1f6981249f975
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>;tag=as1f32ff34
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 33612 ACK
User-Agent: Grandstream HT486 1.0.5.10
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
11 headers, 0 lines
uneu*CLI>
Sip read:
INVITE sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bKfc13585f805afee6
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>
Contact: <sip:[email protected]>
Proxy-Authorization: DIGEST username="70", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", nonce="167fb37c", response="7d1230084e80137e910e0171a53be6e2"
Call-ID: [email protected]
CSeq: 33613 INVITE
User-Agent: Grandstream HT486 1.0.5.10
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 315
v=0
o=70 8000 8000 IN IP4 213.54.212.41
s=SIP Call
c=IN IP4 213.54.212.41
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 2 15 4
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:4 G723/8000
a=ptime:40
13 headers, 15 lines
Using latest request as basis request
Sending to 213.54.212.41 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found audio format ULAW
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G726-32
Found description format G728
Found description format G723
Capabilities: us - 1030, them - 1309/0, combined - 1028
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for *203416894937 in monsterhase
list_route: hop: <sip:[email protected]>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bKfc13585f805afee6
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>;tag=as0f6499c0
Call-ID: [email protected]
CSeq: 33613 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*[email protected]>
Content-Length: 0
to 213.54.212.41:5060
-- parse_srv: SRV mapped to host sipsnip.com, port 5060
We're at 217.20.120.121 port 14588
Answering with preferred capability 2
Answering with preferred capability 4
Answering with preferred capability 1024
Answering with non-codec capability 1
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 217.20.120.121:5060;branch=z9hG4bK2dcc351e
From: "70" <sip:[email protected]>;tag=as0c8f41f4
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 11 Aug 2004 20:03:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 15602 15602 IN IP4 217.20.120.121
s=session
c=IN IP4 217.20.120.121
t=0 0
m=audio 14588 RTP/AVP 3 0 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 217.160.220.106:5060
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bKfc13585f805afee6
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>;tag=as0f6499c0
Call-ID: [email protected]
CSeq: 33613 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*[email protected]>
Content-Length: 0
to 213.54.212.41:5060
uneu*CLI>
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.20.120.121:5060;branch=z9hG4bK2dcc351e
From: "70" <sip:[email protected]>;tag=as0c8f41f4
To: <sip:[email protected]>;tag=4dbcba0cb490d1ab41a8995211c1ac07.0bf1
Call-ID: [email protected]
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="sipsnip.com", nonce="411a7cd034b9d20302cab2f5f53095558e959a26"
Server: Sip EXpress router (0.8.12-1rc1 (i386/linux))
Content-Length: 0
Warning: 392 sipsnip.com:5060 "Noisy feedback tells: pid=4453 req_src_ip=217.20.120.121 req_src_port=5060 in_uri=sip:[email protected] out_uri=sip:[email protected] via_cnt==1"
10 headers, 0 lines
Transmitting:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 217.20.120.121:5060;branch=z9hG4bK2dcc351e
From: "70" <sip:[email protected]>;tag=as0c8f41f4
To: <sip:[email protected]>;tag=4dbcba0cb490d1ab41a8995211c1ac07.0bf1
Contact: <sip:[email protected]>
Call-ID: [email protected]0.121
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 217.160.220.106:5060
Aug 11 22:03:57 NOTICE[4101]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"70" <sip:[email protected]>;tag=as0c8f41f4'
uneu*CLI>
Sip read:
CANCEL sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bKfc13585f805afee6
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>
Contact: <sip:[email protected]>
Proxy-Authorization: DIGEST username="70", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", nonce="167fb37c", response="cd3ffa35f90e2f53b3b17f05c3757ada"
Call-ID: [email protected]
CSeq: 33613 CANCEL
User-Agent: Grandstream HT486 1.0.5.10
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
12 headers, 0 lines
Sending to 213.54.212.41 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bKfc13585f805afee6
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>;tag=as0f6499c0
Call-ID: [email protected]
CSeq: 33613 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*[email protected]>
Content-Length: 0
to 213.54.212.41:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bKfc13585f805afee6
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>;tag=as0f6499c0
Call-ID: [email protected]
CSeq: 33613 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*[email protected]>
Content-Length: 0
to 213.54.212.41:5060
uneu*CLI>
Sip read:
ACK sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bKfc13585f805afee6
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>;tag=as0f6499c0
Contact: <sip:[email protected]>
Proxy-Authorization: DIGEST username="70", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", nonce="167fb37c", response="cd3ffa35f90e2f53b3b17f05c3757ada"
Call-ID: [email protected]
CSeq: 33613 ACK
User-Agent: Grandstream HT486 1.0.5.10
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
12 headers, 0 lines
uneu*CLI>
Sip read:
ACK sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 213.54.212.41;branch=z9hG4bK3431ed5367c8b005
From: <sip:[email protected]>;tag=6c6f475c9ca1ec8a
To: <sip:*[email protected]>;tag=as0f6499c0
Contact: <sip:[email protected]>
Proxy-Authorization: DIGEST username="70", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", nonce="167fb37c", response="6268cb311dd5afdf7c4d1a2b341f20b1"
Call-ID: [email protected]
CSeq: 33613 ACK
User-Agent: Grandstream HT486 1.0.5.10
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
12 headers, 0 lines