rendered paste body
*CLI>
<--- SIP read from UDP:192.168.10.33:5065 --->
INVITE sip:123@192.168.10.43 SIP/2.0
Date: Mon, 06 Jun 2011 07:01:27 GMT
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bK32f97d77-788e-e011-9ec0-d48564eb06ac;rport
User-Agent: Ekiga/2.0.12
From: "max lit" <sip:litnimax@192.168.10.43>;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac
Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer
To: <sip:123@192.168.10.43>
Contact: <sip:litnimax@192.168.10.33:5062;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Content-Length: 206
Max-Forwards: 70
v=0
o=- 1307343687 1307343687 IN IP4 192.168.10.33
s=Opal SIP Session
c=IN IP4 192.168.10.33
t=0 0
m=audio 5004 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 9 lines) ---
Sending to 192.168.10.33:5065 (no NAT)
Using INVITE request as basis request - 24084b77-788e-e011-9ec0-d48564eb06ac@explorer
Found peer '192.168.10.33' for 'litnimax' from 192.168.10.33:5065
<--- Reliably Transmitting (no NAT) to 192.168.10.33:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bK32f97d77-788e-e011-9ec0-d48564eb06ac;received=192.168.10.33;rport=5065
From: "max lit" <sip:litnimax@192.168.10.43>;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac
To: <sip:123@192.168.10.43>;tag=as1ebd5092
Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5db2c25b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '24084b77-788e-e011-9ec0-d48564eb06ac@explorer' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.10.33:5065 --->
ACK sip:123@192.168.10.43 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bK32f97d77-788e-e011-9ec0-d48564eb06ac;rport
From: "max lit" <sip:litnimax@192.168.10.43>;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac
Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer
To: <sip:123@192.168.10.43>;tag=as1ebd5092
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Length: 0
Max-Forwards: 70
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:192.168.10.33:5065 --->
INVITE sip:123@192.168.10.43 SIP/2.0
Date: Mon, 06 Jun 2011 07:01:27 GMT
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bKca327f77-788e-e011-9ec0-d48564eb06ac;rport
User-Agent: Ekiga/2.0.12
Authorization: Digest username="192.168.10.33", realm="asterisk", nonce="5db2c25b", uri="sip:123@192.168.10.43", algorithm=md5, response="c97071284031081c32b827927bcf0482"
From: "max lit" <sip:litnimax@192.168.10.43>;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac
Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer
To: <sip:123@192.168.10.43>
Contact: <sip:litnimax@192.168.10.33:5062;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Content-Length: 206
Max-Forwards: 70
v=0
o=- 1307343687 1307343687 IN IP4 192.168.10.33
s=Opal SIP Session
c=IN IP4 192.168.10.33
t=0 0
m=audio 5004 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 9 lines) ---
Sending to 192.168.10.33:5065 (no NAT)
Using INVITE request as basis request - 24084b77-788e-e011-9ec0-d48564eb06ac@explorer
Found peer '192.168.10.33' for '192.168.10.33' from 192.168.10.33:5065
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.10.33:5004
Looking for 123 in from-max (domain 192.168.10.43)
<--- Reliably Transmitting (no NAT) to 192.168.10.33:5065 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bKca327f77-788e-e011-9ec0-d48564eb06ac;received=192.168.10.33;rport=5065
From: "max lit" <sip:litnimax@192.168.10.43>;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac
To: <sip:123@192.168.10.43>;tag=as1ebd5092
Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.4.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Jun 6 11:00:53] NOTICE[22766]: chan_sip.c:21619 handle_request_invite: Call from 'maxauth' to extension '123' rejected because extension not found in context 'from-max'.
Scheduling destruction of SIP dialog '24084b77-788e-e011-9ec0-d48564eb06ac@explorer' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.10.33:5065 --->
ACK sip:123@192.168.10.43 SIP/2.0
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bKca327f77-788e-e011-9ec0-d48564eb06ac;rport
Authorization: Digest username="192.168.10.33", realm="asterisk", nonce="5db2c25b", uri="sip:123@192.168.10.43", algorithm=md5, response="0d24bb0b38c42908bc292abd66a94cb8"
From: "max lit" <sip:litnimax@192.168.10.43>;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac
Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer
To: <sip:123@192.168.10.43>;tag=as1ebd5092
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Length: 0
Max-Forwards: 70
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 'de290942-788e-e011-9ec0-d48564eb06ac@explorer' Method: REGISTER
Really destroying SIP dialog '24084b77-788e-e011-9ec0-d48564eb06ac@explorer' Method: ACK