Central1_

Parsing /etc/asterisk/asterisk.conf
Seeding global EID '00:01:6c:c7:7f:a3' from 'eth0' using 'siocgifhwaddr'
Asterisk 1.8.21.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
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.
=========================================================================
Connected to Asterisk 1.8.21.0 currently running on home (pid = 1293)
home*CLI> ^M^[[0KVerbosity is at least 32
Core debug was 0 and is now 2
^M^[[Khome*CLI> ^M^[[0K  == Using SIP RTP CoS mark 5
[Apr 29 13:06:21] ^[[1;31mWARNING^[[0m[1603]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m10067^[[0m ^[[1;37mprocess_sdp_a_audio^[[0m: Got Siren7 offer at 24000 bps, but only 3200$
^M^[[Khome*CLI> ^M^[[0K    -- Executing [3000@default:1] ^[[1;36mDial^[[0m("^[[1;35mSIP/2001-00000000^[[0m", "^[[1;35msip/central2/3000,30,tTM^[[0m") in new stack
^M^[[Khome*CLI> ^M^[[0K  == Using SIP RTP CoS mark 5
^M^[[Khome*CLI> ^M^[[0K    -- Called sip/central2/3000
^M^[[Khome*CLI> ^M^[[0K[Apr 29 13:06:21] ^[[1;33mNOTICE^[[0m[1603]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m20959^[[0m ^[[1;37mhandle_response_invite^[[0m: Failed to authentic$
^M^[[Khome*CLI> ^M^[[0K    -- SIP/central2-00000001 is circuit-busy
^M^[[Khome*CLI> ^M^[[0K  == Everyone is busy/congested at this time (1:0/1/0)
^M^[[Khome*CLI> ^M^[[0K    -- Auto fallthrough, channel 'SIP/2001-00000000' status is 'CONGESTION'
^M^[[Khome*CLI> exit^H^H^H^Hsip show peers^[[11Gexit^[[K^H^H^H^Hsip show peers^[[11Gexit^[[K^H^H^H^Hsip show peers^[[11Gexit^[[K^H^H^H^Hsip set debug off^H^Hn^[[K
home*CLI> ^M^[[0KSIP Debugging enabled
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj3e5da521beb745eca75cc4423e990933
Max-Forwards: 70
From: sip:[email protected];tag=0c1ed711537c40d7b5e1a10edcf4bea5
To: sip:[email protected]
Contact: <sip:2001@192.168.1.4:5070>
Call-ID: e0c3df523c134c74b8f2c292fc30cb95
CSeq: 23048 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: AdoreSoftphone
Content-Type: application/sdp
Content-Length: 729
v=0
o=- 3576253913 3576253913 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4003 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 30 lines) ---
Sending to 192.168.1.4:5070 (NAT)
Using INVITE request as basis request - e0c3df523c134c74b8f2c292fc30cb95
Found peer '2001' for '2001' from 192.168.1.4:5070

<--- Reliably Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj3e5da521beb745eca75cc4423e990933;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=0c1ed711537c40d7b5e1a10edcf4bea5
To: sip:[email protected];tag=as690e818d
Call-ID: e0c3df523c134c74b8f2c292fc30cb95
CSeq: 23048 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17a68417"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e0c3df523c134c74b8f2c292fc30cb95' in 6400 ms (Method: INVITE)
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj3e5da521beb745eca75cc4423e990933
Max-Forwards: 70
From: sip:[email protected];tag=0c1ed711537c40d7b5e1a10edcf4bea5
To: sip:[email protected];tag=as690e818d
Call-ID: e0c3df523c134c74b8f2c292fc30cb95
CSeq: 23048 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj4e081d0dc6fd4739ac7f7ad54c0398ee
Max-Forwards: 70
From: sip:[email protected];tag=0c1ed711537c40d7b5e1a10edcf4bea5
To: sip:[email protected]
Contact: <sip:2001@192.168.1.4:5070>
Call-ID: e0c3df523c134c74b8f2c292fc30cb95
CSeq: 23049 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: AdoreSoftphone
Authorization: Digest username="2001", realm="asterisk", nonce="17a68417", uri="sip:[email protected]", response="1df4be9053e771794237ff1c755113a7", algorithm=MD5
Content-Type: application/sdp
Content-Length: 729

v=0
o=- 3576253913 3576253913 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4003 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
^M^[[Khome*CLI> ^M^[[0K--- (14 headers 30 lines) ---
Sending to 192.168.1.4:5070 (NAT)
Using INVITE request as basis request - e0c3df523c134c74b8f2c292fc30cb95
Found peer '2001' for '2001' from 192.168.1.4:5070
  == Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 117
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 118
Found RTP audio format 119
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 125
Found RTP audio format 126
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found unknown media description format speex for ID 104
Found audio description format iLBC for ID 117
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found unknown media description format AMR for ID 118
Found unknown media description format AMR-WB for ID 119
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format G726-32 for ID 2
Found unknown media description format G728 for ID 15
Found audio description format G7221 for ID 125
[Apr 29 13:06:50] ^[[1;31mWARNING^[[0m[1603]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m10067^[[0m ^[[1;37mprocess_sdp_a_audio^[[0m: Got Siren7 offer at 24000 bps, but only 3200$
Found audio description format G7221 for ID 126
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x200003f0f (g723|gsm|ulaw|alaw|g726|g729|speex|speex16|ilbc|g722|siren7)/video=0x0 (nothi$
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.4:4002
Looking for 3000 in default (domain 192.168.1.201)
list_route: hop: <sip:2001@192.168.1.4:5070>

<--- Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj4e081d0dc6fd4739ac7f7ad54c0398ee;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=0c1ed711537c40d7b5e1a10edcf4bea5
To: sip:[email protected]
Call-ID: e0c3df523c134c74b8f2c292fc30cb95
CSeq: 23049 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3000@192.168.1.201:5060>
Content-Length: 0


<------------>
^M^[[Khome*CLI> ^M^[[0K    -- Executing [3000@default:1] ^[[1;36mDial^[[0m("^[[1;35mSIP/2001-00000002^[[0m", "^[[1;35msip/central2/3000,30,tTM^[[0m") in new stack
^M^[[Khome*CLI> ^M^[[0K  == Using SIP RTP CoS mark 5
^M^[[Khome*CLI> ^M^[[0KAudio is at 19342
^M^[[Khome*CLI> ^M^[[0KAdding codec 0x4 (ulaw) to SDP
^M^[[Khome*CLI> ^M^[[0KAdding codec 0x2 (gsm) to SDP
^M^[[Khome*CLI> ^M^[[0KAdding non-codec 0x1 (telephone-event) to SDP
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.202:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK0b161d10;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.201>;tag=as52f3269a
To: <sip:3000@192.168.1.202>
Contact: <sip:2001@192.168.1.201:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:06:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1123005417 1123005417 IN IP4 192.168.1.201
s=Asterisk PBX 1.8.21.0
c=IN IP4 192.168.1.201
t=0 0
m=audio 19342 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
^M^[[Khome*CLI> ^M^[[0K    -- Called sip/central2/3000
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK0b161d10;received=192.168.1.201;rport=5060
From: "2001" <sip:2001@192.168.1.201>;tag=as52f3269a
To: <sip:3000@192.168.1.202>;tag=as30a5c3e0
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="240d42bf"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.202:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK0b161d10;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.201>;tag=as52f3269a
To: <sip:3000@192.168.1.202>;tag=as30a5c3e0
Contact: <sip:2001@192.168.1.201:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.21.0
Content-Length: 0


---
[Apr 29 13:06:50] ^[[1;33mNOTICE^[[0m[1603]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m20959^[[0m ^[[1;37mhandle_response_invite^[[0m: Failed to authenticate on INVITE to '"2001$
^M^[[Khome*CLI> ^M^[[0K    -- SIP/central2-00000003 is circuit-busy
^M^[[Khome*CLI> ^M^[[0K  == Everyone is busy/congested at this time (1:0/1/0)
^M^[[Khome*CLI> ^M^[[0K    -- Auto fallthrough, channel 'SIP/2001-00000002' status is 'CONGESTION'
^M^[[Khome*CLI> ^M^[[0K
<--- Reliably Transmitting (NAT) to 192.168.1.4:5070 --->

SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj4e081d0dc6fd4739ac7f7ad54c0398ee;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=0c1ed711537c40d7b5e1a10edcf4bea5
To: sip:[email protected];tag=as354be6d3
Call-ID: e0c3df523c134c74b8f2c292fc30cb95
CSeq: 23049 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: INVITE
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj4e081d0dc6fd4739ac7f7ad54c0398ee
Max-Forwards: 70
From: sip:[email protected];tag=0c1ed711537c40d7b5e1a10edcf4bea5
To: sip:[email protected];tag=as354be6d3
Call-ID: e0c3df523c134c74b8f2c292fc30cb95
CSeq: 23049 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'e0c3df523c134c74b8f2c292fc30cb95' Method: ACK
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->

<------------->
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.4:5070:
OPTIONS sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK39c7ab61;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1f5c043d
To: <sip:2001@192.168.1.4:5070>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:06:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.201;branch=z9hG4bK39c7ab61
Call-ID: [email protected]:5060
From: "asterisk" <sip:[email protected]>;tag=as1f5c043d
To: <sip:2001@192.168.1.4>
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, norefersub
Allow-Events: presence, refer
User-Agent: AdoreSoftphone
Content-Type: application/sdp
Content-Length: 718

v=0
o=- 3576253914 3576253914 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4001 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 29 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (no NAT) to 200.43.153.237:5060:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK2254c8b8
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as28f76935
To: <sip:1000@200.43.153.237>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:06:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:200.43.153.237:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK2254c8b8;received=190.122.72.253
From: "asterisk" <sip:[email protected]>;tag=as28f76935
To: <sip:1000@200.43.153.237>;tag=as2f728dea
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: easyCall (by easyIP)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:200.43.153.237>
Accept: application/sdp
Content-Length: 0

<------------->
^M^[[Khome*CLI> ^M^[[0K--- (12 headers 0 lines) ---
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjb3e652ef23c34e919128655b59becb10
Max-Forwards: 70
From: sip:[email protected];tag=914686e98f1a4df8999a578bfbf45c99
To: sip:[email protected]
Contact: <sip:2001@192.168.1.4:5070>
Call-ID: ea0383d278824b728d8b973eccbc25c9
CSeq: 15615 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: AdoreSoftphone
Content-Type: application/sdp
Content-Length: 729

v=0
o=- 3576253920 3576253920 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
a=X-nat:0
m=audio 4004 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4005 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
^M^[[Khome*CLI> ^M^[[0K--- (13 headers 30 lines) ---
^M^[[Khome*CLI> ^M^[[0KSending to 192.168.1.4:5070 (NAT)
^M^[[Khome*CLI> ^M^[[0KUsing INVITE request as basis request - ea0383d278824b728d8b973eccbc25c9
^M^[[Khome*CLI> ^M^[[0KFound peer '2001' for '2001' from 192.168.1.4:5070
^M^[[Khome*CLI> ^M^[[0K
<--- Reliably Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjb3e652ef23c34e919128655b59becb10;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=914686e98f1a4df8999a578bfbf45c99
To: sip:[email protected];tag=as306bf865
Call-ID: ea0383d278824b728d8b973eccbc25c9
CSeq: 15615 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10379218"
Content-Length: 0
<------------>
^M^[[Khome*CLI> ^M^[[0KScheduling destruction of SIP dialog 'ea0383d278824b728d8b973eccbc25c9' in 6400 ms (Method: INVITE)
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjb3e652ef23c34e919128655b59becb10
Max-Forwards: 70
From: sip:[email protected];tag=914686e98f1a4df8999a578bfbf45c99
To: sip:[email protected];tag=as306bf865
Call-ID: ea0383d278824b728d8b973eccbc25c9
CSeq: 15615 ACK
Content-Length: 0

<------------->
^M^[[Khome*CLI> ^M^[[0K--- (8 headers 0 lines) ---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj60ef846fe81f4518a65635d926a7214b
Max-Forwards: 70

From: sip:[email protected];tag=914686e98f1a4df8999a578bfbf45c99
To: sip:[email protected]
Contact: <sip:2001@192.168.1.4:5070>
Call-ID: ea0383d278824b728d8b973eccbc25c9
CSeq: 15616 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: AdoreSoftphone
Authorization: Digest username="2001", realm="asterisk", nonce="10379218", uri="sip:[email protected]", response="e58ccbb3fe6f0269064553ab0e0f0319", algorithm=MD5
Content-Type: application/sdp
Content-Length: 729

v=0
o=- 3576253920 3576253920 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
a=X-nat:0
m=audio 4004 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4005 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
^M^[[Khome*CLI> ^M^[[0K--- (14 headers 30 lines) ---
^M^[[Khome*CLI> ^M^[[0KSending to 192.168.1.4:5070 (NAT)
^M^[[Khome*CLI> ^M^[[0KUsing INVITE request as basis request - ea0383d278824b728d8b973eccbc25c9
^M^[[Khome*CLI> ^M^[[0KFound peer '2001' for '2001' from 192.168.1.4:5070
^M^[[Khome*CLI> ^M^[[0K  == Using SIP RTP CoS mark 5
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 103
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 102
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 104
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 117
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 3
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 0
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 8
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 9
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 118
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 119
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 18
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 4
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 2
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 15
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 125
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 126
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 101
^M^[[Khome*CLI> ^M^[[0KFound audio description format speex for ID 103
^M^[[Khome*CLI> ^M^[[0KFound audio description format speex for ID 102
^M^[[Khome*CLI> ^M^[[0KFound unknown media description format speex for ID 104
^M^[[Khome*CLI> ^M^[[0KFound audio description format iLBC for ID 117
^M^[[Khome*CLI> ^M^[[0KFound audio description format GSM for ID 3
^M^[[Khome*CLI> ^M^[[0KFound audio description format PCMU for ID 0
^M^[[Khome*CLI> ^M^[[0KFound audio description format PCMA for ID 8
^M^[[Khome*CLI> ^M^[[0KFound audio description format G722 for ID 9
^M^[[Khome*CLI> ^M^[[0KFound unknown media description format AMR for ID 118
^M^[[Khome*CLI> ^M^[[0KFound unknown media description format AMR-WB for ID 119
^M^[[Khome*CLI> ^M^[[0KFound audio description format G729 for ID 18
^M^[[Khome*CLI> ^M^[[0KFound audio description format G723 for ID 4
^M^[[Khome*CLI> ^M^[[0KFound audio description format G726-32 for ID 2
^M^[[Khome*CLI> ^M^[[0KFound unknown media description format G728 for ID 15
^M^[[Khome*CLI> ^M^[[0KFound audio description format G7221 for ID 125
^M^[[Khome*CLI> ^M^[[0K[Apr 29 13:06:57] ^[[1;31mWARNING^[[0m[1603]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m10067^[[0m ^[[1;37mprocess_sdp_a_audio^[[0m: Got Siren7 offer at 2$
^M^[[Khome*CLI> ^M^[[0KFound audio description format G7221 for ID 126
^M^[[Khome*CLI> ^M^[[0KFound audio description format telephone-event for ID 101
^M^[[Khome*CLI> ^M^[[0KCapabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x200003f0f (g723|gsm|ulaw|alaw|g726|g729|speex|speex16|ilbc|g722|s$
^M^[[Khome*CLI> ^M^[[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
^M^[[Khome*CLI> ^M^[[0KPeer audio RTP is at port 192.168.1.4:4004
^M^[[Khome*CLI> ^M^[[0KLooking for 3000 in default (domain 192.168.1.201)
^M^[[Khome*CLI> ^M^[[0Klist_route: hop: <sip:2001@192.168.1.4:5070>
^M^[[Khome*CLI> ^M^[[0K
<--- Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj60ef846fe81f4518a65635d926a7214b;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=914686e98f1a4df8999a578bfbf45c99
To: sip:[email protected]
Call-ID: ea0383d278824b728d8b973eccbc25c9
CSeq: 15616 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3000@192.168.1.201:5060>
Content-Length: 0


<------------>
^M^[[Khome*CLI> ^M^[[0K    -- Executing [3000@default:1] ^[[1;36mDial^[[0m("^[[1;35mSIP/2001-00000004^[[0m", "^[[1;35msip/central2/3000,30,tTM^[[0m") in new stack
^M^[[Khome*CLI> ^M^[[0K  == Using SIP RTP CoS mark 5
^M^[[Khome*CLI> ^M^[[0KAudio is at 12906
^M^[[Khome*CLI> ^M^[[0KAdding codec 0x4 (ulaw) to SDP
^M^[[Khome*CLI> ^M^[[0KAdding codec 0x2 (gsm) to SDP
^M^[[Khome*CLI> ^M^[[0KAdding non-codec 0x1 (telephone-event) to SDP
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.202:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK78b4b886;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.201>;tag=as4d1e93e3
To: <sip:3000@192.168.1.202>
Contact: <sip:2001@192.168.1.201:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:06:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1619176956 1619176956 IN IP4 192.168.1.201
s=Asterisk PBX 1.8.21.0
c=IN IP4 192.168.1.201
t=0 0
m=audio 12906 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
---
^M^[[Khome*CLI> ^M^[[0K    -- Called sip/central2/3000
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK78b4b886;received=192.168.1.201;rport=5060
From: "2001" <sip:2001@192.168.1.201>;tag=as4d1e93e3
To: <sip:3000@192.168.1.202>;tag=as79ad0df9
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a5cadaf"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.202:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK78b4b886;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.201>;tag=as4d1e93e3
To: <sip:3000@192.168.1.202>;tag=as79ad0df9
Contact: <sip:2001@192.168.1.201:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.21.0
Content-Length: 0


---
[Apr 29 13:06:57] ^[[1;33mNOTICE^[[0m[1603]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m20959^[[0m ^[[1;37mhandle_response_invite^[[0m: Failed to authenticate on INVITE to '"2001$
^M^[[Khome*CLI> ^M^[[0K    -- SIP/central2-00000005 is circuit-busy
^M^[[Khome*CLI> ^M^[[0K  == Everyone is busy/congested at this time (1:0/1/0)
^M^[[Khome*CLI> ^M^[[0K    -- Auto fallthrough, channel 'SIP/2001-00000004' status is 'CONGESTION'
^M^[[Khome*CLI> ^M^[[0K
<--- Reliably Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj60ef846fe81f4518a65635d926a7214b;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=914686e98f1a4df8999a578bfbf45c99
To: sip:[email protected];tag=as7547904a
Call-ID: ea0383d278824b728d8b973eccbc25c9
CSeq: 15616 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: INVITE
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj60ef846fe81f4518a65635d926a7214b
Max-Forwards: 70
From: sip:[email protected];tag=914686e98f1a4df8999a578bfbf45c99
To: sip:[email protected];tag=as7547904a
Call-ID: ea0383d278824b728d8b973eccbc25c9
CSeq: 15616 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'ea0383d278824b728d8b973eccbc25c9' Method: ACK
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK54e6d1ed;rport
Max-Forwards: 70
From: "3000" <sip:3000@192.168.1.202>;tag=as159b7043
To: <sip:2001@192.168.1.201>
Contact: <sip:3000@192.168.1.202:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 19:52:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1934654958 1934654958 IN IP4 192.168.1.202
s=Asterisk PBX 1.8.21.0
c=IN IP4 192.168.1.202
t=0 0
m=audio 19040 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.202:5060 (NAT)
Using INVITE request as basis request - [email protected]:5060
Found peer 'central2' for '3000' from 192.168.1.202:5060

<--- Reliably Transmitting (NAT) to 192.168.1.202:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK54e6d1ed;received=192.168.1.202;rport=5060
From: "3000" <sip:3000@192.168.1.202>;tag=as159b7043
To: <sip:2001@192.168.1.201>;tag=as17b3a9e2
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="180145f6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK54e6d1ed;rport
Max-Forwards: 70
From: "3000" <sip:3000@192.168.1.202>;tag=as159b7043
To: <sip:2001@192.168.1.201>;tag=as17b3a9e2
Contact: <sip:3000@192.168.1.202:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.21.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5060 --->


<------------->
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->

<------------->
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: ACK
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.202:5060:
OPTIONS sip:192.168.1.202 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK01896e29;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as5fe491d1
To: <sip:192.168.1.202>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:07:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK01896e29;received=192.168.1.201;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as5fe491d1
To: <sip:192.168.1.202>;tag=as2e846901
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.1.202:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> ^M^[[0K

<--- SIP read from UDP:192.168.1.4:5070 --->

<------------->
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
OPTIONS sip:192.168.1.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7216ad02;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as095c061a
To: <sip:192.168.1.201>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 19:52:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in default (domain 192.168.1.201)

<--- Transmitting (NAT) to 192.168.1.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7216ad02;received=192.168.1.202;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as095c061a
To: <sip:192.168.1.201>;tag=as50deca58
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.1.201:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: OPTIONS)
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5060 --->


<------------->
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.4:5060:
OPTIONS sip:[email protected]:5060;rinstance=3fe875f12fb1b287;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK45af0122;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as4e51cff0
To: <sip:2000@190.122.72.253:5060;rinstance=3fe875f12fb1b287;transport=UDP>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:07:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK45af0122;rport=5060
Contact: <sip:192.168.1.4:5060>
To: <sip:2000@190.122.72.253:5060;rinstance=3fe875f12fb1b287;transport=UDP>;tag=234f1f42
From: "asterisk"<sip:[email protected]>;tag=as4e51cff0
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->

<------------->
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
REGISTER sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj2ec245a2052740a99da3622e822cd5c3
Max-Forwards: 70
From: <sip:2001@192.168.1.201>;tag=e9196f0111e0442a9bd5acd1d02c8c37
To: <sip:2001@192.168.1.201>
Call-ID: 600ce7ed39624b2a8bc32daeb5a7c412
CSeq: 51948 REGISTER
User-Agent: AdoreSoftphone
Contact: <sip:2001@192.168.1.4:5070>
Expires: 300
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.4:5070 (NAT)

<--- Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj2ec245a2052740a99da3622e822cd5c3;received=192.168.1.4;rport=5070
From: <sip:2001@192.168.1.201>;tag=e9196f0111e0442a9bd5acd1d02c8c37
To: <sip:2001@192.168.1.201>;tag=as04492cc6
Call-ID: 600ce7ed39624b2a8bc32daeb5a7c412
CSeq: 51948 REGISTER
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a5d9d34"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '600ce7ed39624b2a8bc32daeb5a7c412' in 32000 ms (Method: REGISTER)
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
REGISTER sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj30aecb8264bb47ef95650bb15686cf95
Max-Forwards: 70
From: <sip:2001@192.168.1.201>;tag=e9196f0111e0442a9bd5acd1d02c8c37
To: <sip:2001@192.168.1.201>
Call-ID: 600ce7ed39624b2a8bc32daeb5a7c412
CSeq: 51949 REGISTER
User-Agent: AdoreSoftphone
Contact: <sip:2001@192.168.1.4:5070>
Expires: 300
Authorization: Digest username="2001", realm="asterisk", nonce="2a5d9d34", uri="sip:[email protected]", response="e0f3c2ae27059e2dbc88e930700edce4", algorithm=MD5
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.4:5070 (NAT)
Reliably Transmitting (NAT) to 192.168.1.4:5070:
OPTIONS sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK7d208055;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1d232d1a
To: <sip:2001@192.168.1.4:5070>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:07:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj30aecb8264bb47ef95650bb15686cf95;received=192.168.1.4;rport=5070
From: <sip:2001@192.168.1.201>;tag=e9196f0111e0442a9bd5acd1d02c8c37
To: <sip:2001@192.168.1.201>;tag=as04492cc6
Call-ID: 600ce7ed39624b2a8bc32daeb5a7c412
CSeq: 51949 REGISTER
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:2001@192.168.1.4:5070>;expires=300
Date: Mon, 29 Apr 2013 16:07:46 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.4:5070:
NOTIFY sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK42eb51d9;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as2abc22cc
To: <sip:2001@192.168.1.4:5070>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.21.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:[email protected]
Voice-Message: 0/0 (0/0)

---
^M^[[Khome*CLI> ^M^[[0KScheduling destruction of SIP dialog '600ce7ed39624b2a8bc32daeb5a7c412' in 32000 ms (Method: REGISTER)
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.201;branch=z9hG4bK7d208055
Call-ID: [email protected]:5060
From: "asterisk" <sip:[email protected]>;tag=as1d232d1a
To: <sip:2001@192.168.1.4>
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, norefersub
Allow-Events: presence, refer
User-Agent: AdoreSoftphone
Content-Type: application/sdp
Content-Length: 718

v=0
o=- 3576253969 3576253969 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4001 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
^M^[[Khome*CLI> ^M^[[0K--- (13 headers 29 lines) ---
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.201;branch=z9hG4bK42eb51d9
Call-ID: [email protected]:5060
From: "asterisk" <sip:[email protected]>;tag=as2abc22cc
To: <sip:2001@192.168.1.4>
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
^M^[[Khome*CLI> ^M^[[0K--- (7 headers 0 lines) ---
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: NOTIFY
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK00c124db;rport
Max-Forwards: 70
From: "3000" <sip:3000@192.168.1.202>;tag=as221104e9
To: <sip:2001@192.168.1.201>
Contact: <sip:3000@192.168.1.202:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 19:52:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1848754928 1848754928 IN IP4 192.168.1.202
s=Asterisk PBX 1.8.21.0
c=IN IP4 192.168.1.202
t=0 0
m=audio 19026 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
^M^[[Khome*CLI> ^M^[[0K--- (14 headers 12 lines) ---
^M^[[Khome*CLI> ^M^[[0KSending to 192.168.1.202:5060 (NAT)
^M^[[Khome*CLI> ^M^[[0KUsing INVITE request as basis request - [email protected]:5060
^M^[[Khome*CLI> ^M^[[0KFound peer 'central2' for '3000' from 192.168.1.202:5060
^M^[[Khome*CLI> ^M^[[0K
<--- Reliably Transmitting (NAT) to 192.168.1.202:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK00c124db;received=192.168.1.202;rport=5060
From: "3000" <sip:3000@192.168.1.202>;tag=as221104e9
To: <sip:2001@192.168.1.201>;tag=as5caa7d00
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="55bd9d54"
Content-Length: 0


<------------>
^M^[[Khome*CLI> ^M^[[0KScheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK00c124db;rport
Max-Forwards: 70
From: "3000" <sip:3000@192.168.1.202>;tag=as221104e9
To: <sip:2001@192.168.1.201>;tag=as5caa7d00
Contact: <sip:3000@192.168.1.202:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.21.0
Content-Length: 0

<------------->
^M^[[Khome*CLI> ^M^[[0K--- (10 headers 0 lines) ---
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: ACK
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (no NAT) to 200.43.153.237:5060:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK6cf7e106
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as01fdbb43
To: <sip:1000@200.43.153.237>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:07:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:200.43.153.237:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK6cf7e106;received=190.122.72.253
From: "asterisk" <sip:[email protected]>;tag=as01fdbb43
To: <sip:1000@200.43.153.237>;tag=as5ca8e439
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: easyCall (by easyIP)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:200.43.153.237>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjbfdb0e6e81c94803baab5303deea9825
Max-Forwards: 70
From: sip:[email protected];tag=822e23cde6ac41dead279234ec9f9b5c
To: sip:[email protected]
Contact: <sip:2001@192.168.1.4:5070>
Call-ID: 78232eed06e145cdae82489eca71cdd9
CSeq: 17365 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: AdoreSoftphone
Content-Type: application/sdp
Content-Length: 729

v=0
o=- 3576253981 3576253981 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4007 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 30 lines) ---
Sending to 192.168.1.4:5070 (NAT)
Using INVITE request as basis request - 78232eed06e145cdae82489eca71cdd9
Found peer '2001' for '2001' from 192.168.1.4:5070

<--- Reliably Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjbfdb0e6e81c94803baab5303deea9825;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=822e23cde6ac41dead279234ec9f9b5c
To: sip:[email protected];tag=as29c408eb
Call-ID: 78232eed06e145cdae82489eca71cdd9
CSeq: 17365 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53e4d706"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '78232eed06e145cdae82489eca71cdd9' in 6400 ms (Method: INVITE)
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjbfdb0e6e81c94803baab5303deea9825
Max-Forwards: 70
From: sip:[email protected];tag=822e23cde6ac41dead279234ec9f9b5c
To: sip:[email protected];tag=as29c408eb
Call-ID: 78232eed06e145cdae82489eca71cdd9
CSeq: 17365 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjfe0778b7bbf94411a53a4720b5ba69ff
Max-Forwards: 70
From: sip:[email protected];tag=822e23cde6ac41dead279234ec9f9b5c
To: sip:[email protected]
Contact: <sip:2001@192.168.1.4:5070>
Call-ID: 78232eed06e145cdae82489eca71cdd9
CSeq: 17366 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: AdoreSoftphone
Authorization: Digest username="2001", realm="asterisk", nonce="53e4d706", uri="sip:[email protected]", response="c7e38ac527841272c006f998cff46de8", algorithm=MD5
Content-Type: application/sdp
Content-Length: 729

v=0
o=- 3576253981 3576253981 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4007 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 30 lines) ---
Sending to 192.168.1.4:5070 (NAT)
Using INVITE request as basis request - 78232eed06e145cdae82489eca71cdd9
Found peer '2001' for '2001' from 192.168.1.4:5070
  == Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 117
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 118
Found RTP audio format 119
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 2
^M^[[Khome*CLI> ^M^[[0KFound RTP audio format 15
Found RTP audio format 125
Found RTP audio format 126
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found unknown media description format speex for ID 104
Found audio description format iLBC for ID 117
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found unknown media description format AMR for ID 118
Found unknown media description format AMR-WB for ID 119
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format G726-32 for ID 2
Found unknown media description format G728 for ID 15
Found audio description format G7221 for ID 125
[Apr 29 13:07:58] ^[[1;31mWARNING^[[0m[1603]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m10067^[[0m ^[[1;37mprocess_sdp_a_audio^[[0m: Got Siren7 offer at 24000 bps, but only 3200$
Found audio description format G7221 for ID 126
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x200003f0f (g723|gsm|ulaw|alaw|g726|g729|speex|speex16|ilbc|g722|siren7)/video=0x0 (nothi$
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.4:4006
Looking for 3000 in default (domain 192.168.1.201)
list_route: hop: <sip:2001@192.168.1.4:5070>

<--- Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjfe0778b7bbf94411a53a4720b5ba69ff;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=822e23cde6ac41dead279234ec9f9b5c
To: sip:[email protected]
Call-ID: 78232eed06e145cdae82489eca71cdd9
CSeq: 17366 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3000@192.168.1.201:5060>
Content-Length: 0


<------------>
^M^[[Khome*CLI> ^M^[[0K    -- Executing [3000@default:1] ^[[1;36mDial^[[0m("^[[1;35mSIP/2001-00000006^[[0m", "^[[1;35msip/central2/3000,30,tTM^[[0m") in new stack
^M^[[Khome*CLI> ^M^[[0K  == Using SIP RTP CoS mark 5
^M^[[Khome*CLI> ^M^[[0KAudio is at 13830
^M^[[Khome*CLI> ^M^[[0KAdding codec 0x4 (ulaw) to SDP
^M^[[Khome*CLI> ^M^[[0KAdding codec 0x2 (gsm) to SDP
^M^[[Khome*CLI> ^M^[[0KAdding non-codec 0x1 (telephone-event) to SDP
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.202:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1dc38b11;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.201>;tag=as39c3d829
To: <sip:3000@192.168.1.202>
Contact: <sip:2001@192.168.1.201:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:07:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 700161715 700161715 IN IP4 192.168.1.201
s=Asterisk PBX 1.8.21.0
c=IN IP4 192.168.1.201
t=0 0
m=audio 13830 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
^M^[[Khome*CLI> ^M^[[0K    -- Called sip/central2/3000
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1dc38b11;received=192.168.1.201;rport=5060
From: "2001" <sip:2001@192.168.1.201>;tag=as39c3d829
To: <sip:3000@192.168.1.202>;tag=as4412c3a0
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b2613d"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.202:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1dc38b11;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.201>;tag=as39c3d829
To: <sip:3000@192.168.1.202>;tag=as4412c3a0
Contact: <sip:2001@192.168.1.201:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.21.0
Content-Length: 0


---
[Apr 29 13:07:58] ^[[1;33mNOTICE^[[0m[1603]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m20959^[[0m ^[[1;37mhandle_response_invite^[[0m: Failed to authenticate on INVITE to '"2001$
^M^[[Khome*CLI> ^M^[[0K    -- SIP/central2-00000007 is circuit-busy
^M^[[Khome*CLI> ^M^[[0K  == Everyone is busy/congested at this time (1:0/1/0)
^M^[[Khome*CLI> ^M^[[0K    -- Auto fallthrough, channel 'SIP/2001-00000006' status is 'CONGESTION'
^M^[[Khome*CLI> ^M^[[0K
<--- Reliably Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjfe0778b7bbf94411a53a4720b5ba69ff;received=192.168.1.4;rport=5070
From: sip:[email protected];tag=822e23cde6ac41dead279234ec9f9b5c
To: sip:[email protected];tag=as31b5d937
Call-ID: 78232eed06e145cdae82489eca71cdd9
CSeq: 17366 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: INVITE
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjfe0778b7bbf94411a53a4720b5ba69ff
Max-Forwards: 70
From: sip:[email protected];tag=822e23cde6ac41dead279234ec9f9b5c
To: sip:[email protected];tag=as31b5d937
Call-ID: 78232eed06e145cdae82489eca71cdd9
CSeq: 17366 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '78232eed06e145cdae82489eca71cdd9' Method: ACK
^M^[[Khome*CLI> ^M^[[0KReally destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5070 --->

<------------->
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.4:5060 --->


<------------->
^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.202:5060:
OPTIONS sip:192.168.1.202 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK60f5aae3;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as369cb314
To: <sip:192.168.1.202>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 16:08:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
^M^[[Khome*CLI> ^M^[[0K
<--- SIP read from UDP:192.168.1.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK60f5aae3;received=192.168.1.201;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as369cb314
To: <sip:192.168.1.202>;tag=as4addba28
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.1.202:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
^M^[[Khome*CLI> sip set debug on^H^[[K^H^[[Koff
home*CLI> ^M^[[0KSIP Debugging Disabled
^M^[[Khome*CLI> ei^H^[[Kxit
Executing last minute cleanups
Asterisk ending (0).