rendered paste bodyroot@braserv:~# asterisk -rvvvT
[Jan 31 21:48:25] Asterisk 1.8.8.1, Copyright (C) 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
[Jan 31 21:48:25] == Parsing '/etc/asterisk/asterisk.conf': [Jan 31 21:48:25] == Found
[Jan 31 21:48:25] Connected to Asterisk 1.8.8.1 currently running on braserv (pid = 17303)
Verbosity was 0 and is now 3
braserv*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
casasip/casasip 189.26.255.43 D N 18421 OK (65 ms)
janailton/janailton 192.168.0.103 D N 5060 OK (38 ms)
leandro/leandro 192.168.0.106 D N 5060 OK (24 ms)
ludmila/ludmila 192.168.0.105 D N 5060 OK (67 ms)
rcoffice (Unspecified) D N 0 UNKNOWN
rosana/rosana 192.168.0.104 D N 5060 OK (25 ms)
sasa (Unspecified) D N 0 UNKNOWN
7 sip peers [Monitored: 5 online, 2 offline Unmonitored: 0 online, 0 offline]
== Using SIP RTP CoS mark 5
-- Executing [900@sip-phones:1] VoiceMailMain("SIP/casasip-00000000", "") in new stack
-- <SIP/casasip-00000000> Playing 'vm-login.gsm' (language 'en')
braserv*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:189.26.255.43:18421 --->
INVITE sip:900@braserv.chickenkiller.com SIP/2.0
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bK0bcd64e617ef05ef
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>
Contact: <sip:casasip@192.168.12.99:5060;transport=udp>
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9463 INVITE
User-Agent: Grandstream GXP285 1.2.4.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 409
v=0
o=casasip 8000 8000 IN IP4 192.168.12.99
s=SIP Call
c=IN IP4 192.168.12.99
t=0 0
m=audio 5042 RTP/AVP 0 8 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (14 headers 19 lines) ---
Sending to 189.26.255.43:18421 (NAT)
Using INVITE request as basis request - 1f9d8a4ac7c221f1@192.168.12.99
Found peer 'casasip' for 'casasip' from 189.26.255.43:18421
<--- Reliably Transmitting (NAT) to 189.26.255.43:18421 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bK0bcd64e617ef05ef;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as0685c85f
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9463 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="187421f9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1f9d8a4ac7c221f1@192.168.12.99' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:189.26.255.43:18421 --->
ACK sip:900@braserv.chickenkiller.com SIP/2.0
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bK0bcd64e617ef05ef
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as0685c85f
Contact: <sip:casasip@192.168.12.99:5060;transport=udp>
Supported: path
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9463 ACK
User-Agent: Grandstream GXP285 1.2.4.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:189.26.255.43:18421 --->
INVITE sip:900@braserv.chickenkiller.com SIP/2.0
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>
Contact: <sip:casasip@192.168.12.99:5060;transport=udp>
Supported: replaces, timer, path
P-Early-Media: Supported
Authorization: Digest username="casasip", realm="asterisk", algorithm=MD5, uri="sip:900@braserv.chickenkiller.com", nonce="187421f9", response="24c9e6b0b56dd76179798dbafc8164da"
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
User-Agent: Grandstream GXP285 1.2.4.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 409
v=0
o=casasip 8000 8001 IN IP4 192.168.12.99
s=SIP Call
c=IN IP4 192.168.12.99
t=0 0
m=audio 5042 RTP/AVP 0 8 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (15 headers 19 lines) ---
Sending to 189.26.255.43:18421 (NAT)
Using INVITE request as basis request - 1f9d8a4ac7c221f1@192.168.12.99
Found peer 'casasip' for 'casasip' from 189.26.255.43:18421
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x2 (gsm), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.12.99:5042
Looking for 900 in sip-phones (domain braserv.chickenkiller.com)
list_route: hop: <sip:casasip@192.168.12.99:5060;transport=udp>
<--- Transmitting (NAT) to 189.26.255.43:18421 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:900@192.168.0.1:5060>
Content-Length: 0
<------------>
-- Executing [900@sip-phones:1] VoiceMailMain("SIP/casasip-00000001", "") in new stack
Audio is at 5060
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 189.26.255.43:18421 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:900@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 227
v=0
o=root 6871369 6871369 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.1
t=0 0
m=audio 10064 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 189.26.255.43:18421:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:900@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 227
v=0
o=root 6871369 6871369 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.1
t=0 0
m=audio 10064 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #2 (NAT) to 189.26.255.43:18421:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:900@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 227
v=0
o=root 6871369 6871369 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.1
t=0 0
m=audio 10064 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- <SIP/casasip-00000001> Playing 'vm-login.gsm' (language 'en')
Retransmitting #3 (NAT) to 189.26.255.43:18421:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:900@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 227
v=0
o=root 6871369 6871369 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.1
t=0 0
m=audio 10064 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #4 (NAT) to 189.26.255.43:18421:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:900@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 227
v=0
o=root 6871369 6871369 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.1
t=0 0
m=audio 10064 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #5 (NAT) to 189.26.255.43:18421:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:900@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 227
v=0
o=root 6871369 6871369 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.1
t=0 0
m=audio 10064 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #6 (NAT) to 189.26.255.43:18421:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.99:5060;branch=z9hG4bKf20de686f79d267f;received=189.26.255.43;rport=18421
From: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
To: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 9464 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:900@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 227
v=0
o=root 6871369 6871369 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.1
t=0 0
m=audio 10064 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Scheduling destruction of SIP dialog '1f9d8a4ac7c221f1@192.168.12.99' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:casasip@192.168.12.99:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.12.99:5060
Reliably Transmitting (NAT) to 189.26.255.43:18421:
BYE sip:casasip@192.168.12.99:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK23080af6;rport
Max-Forwards: 70
From: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
To: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.8.1
Proxy-Authorization: Digest username="casasip", realm="asterisk", algorithm=MD5, uri="sip:braserv.chickenkiller.com", nonce="", response="c264dd0e8354640e2db7ea438aab9354"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
---
<--- SIP read from UDP:189.26.255.43:18421 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK23080af6;rport
From: <sip:900@braserv.chickenkiller.com>;tag=as613e6242
To: "Casa Saxa Ludmila" <sip:casasip@braserv.chickenkiller.com>;tag=3fd69fb8efa4a72e
Call-ID: 1f9d8a4ac7c221f1@192.168.12.99
CSeq: 102 BYE
User-Agent: Grandstream GXP285 1.2.4.3
Contact: <sip:casasip@192.168.12.99:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1f9d8a4ac7c221f1@192.168.12.99' Method: INVITE
<--- SIP read from UDP:189.26.255.43:18421 --->
<------------->
braserv*CLI>