rendered paste bodyAsterisk 1.2.17, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.17 currently running on evpp2 (pid = 5804)
Verbosity is at least 25
-- Remote UNIX connection
evpp2*CLI> sip debug
SIP Debugging enabled
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 204.147.183.18:5060:
REGISTER sip:sip.stanaphone.com SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK160c83f2;rport
From: <sip:08157027@sip.stanaphone.com>;tag=as522967e1
To: <sip:08157027@sip.stanaphone.com>
Call-ID: 73bb50220272cd587bc9342a325f9afd@192.168.0.103
CSeq: 712 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="08157027", realm="stanaphone.com", algorithm=MD5, uri="sip:sip.stanaphone.com", nonce="467cd6c68beaebe461ea4ab27000d24e5800acea", response="faeb4d5a9d1805b551aaedfd43c14d04", opaque=""
Expires: 120
Contact: <sip:s@67.161.185.121>
Event: registration
Content-Length: 0
---
evpp2*CLI>
<-- SIP read from 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK160c83f2;rport=5060
From: <sip:08157027@sip.stanaphone.com>;tag=as522967e1
To: <sip:08157027@sip.stanaphone.com>;tag=4e487ad2f2fae333606267674b3a79b1.f7a3
Call-ID: 73bb50220272cd587bc9342a325f9afd@192.168.0.103
CSeq: 712 REGISTER
Contact: <sip:s@67.161.185.121>;expires=115
Server: Sip EXpress router (0.9.5-pre1 (i386/linux))
Content-Length: 0
Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=29580 req_src_ip=67.161.185.121 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1"
--- (10 headers 0 lines) ---
Scheduling destruction of call '73bb50220272cd587bc9342a325f9afd@192.168.0.103' in 32000 ms
-- Starting simple switch on 'Zap/6-1'
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 204.11.192.22:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK7f9eeb61;rport
From: <sip:17772939780@callcentric.com>;tag=as45d872c8
To: <sip:17772939780@callcentric.com>
Call-ID: 6880133c7f613ea67b07b9363907137c@192.168.0.103
CSeq: 1811 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="17772939780", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", nonce="c1b307d390f9be477ebb50fb12e1c808", response="6bb60d2359eeda8cb0db45d878aecf45", opaque=""
Expires: 120
Contact: <sip:s@67.161.185.121>
Event: registration
Content-Length: 0
---
evpp2*CLI>
<-- SIP read from 204.11.192.22:5060:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK7f9eeb61;rport=5060
f: <sip:17772939780@callcentric.com>;tag=as45d872c8
t: <sip:17772939780@callcentric.com>
i: 6880133c7f613ea67b07b9363907137c@192.168.0.103
CSeq: 1811 REGISTER
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="5c5dc3dcde1632cc72fbed4ff915c331", opaque="", stale=TRUE, algorithm=MD5
l: 0
--- (8 headers 0 lines) ---
Responding to challenge, registration to domain/host name callcentric.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 204.11.192.22:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK1985aa19;rport
From: <sip:17772939780@callcentric.com>;tag=as0d2b0762
To: <sip:17772939780@callcentric.com>
Call-ID: 6880133c7f613ea67b07b9363907137c@192.168.0.103
CSeq: 1812 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="17772939780", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", nonce="5c5dc3dcde1632cc72fbed4ff915c331", response="751b58e68591731d74bbfaf53a6374a0", opaque=""
Expires: 120
Contact: <sip:s@67.161.185.121>
Event: registration
Content-Length: 0
---
evpp2*CLI>
<-- SIP read from 204.11.192.22:5060:
NOTIFY sip:s@67.161.185.121 SIP/2.0
v: SIP/2.0/UDP 204.11.192.22:5060;branch=z9hG4bK-d2cce413e7e1e4ed0ec07ce0df0f3667
f: <sip:17772939780@callcentric.com:5062>
t: <sip:17772939780@callcentric.com>
i: 76258d2f8bbc53ea735ef2ebd9a075d6-05f52368f794a3d01c2ea3446313dc66@callcentric.com
CSeq: 1 NOTIFY
Max-Forwards: 15
m: <sip:398c686b3bd5c065f69c67543638ab5a@204.11.192.22:5060;transport=udp>
Event: message-summary
c: application/simple-message-summary
l: 138
essages-Waiting: no
Message-Account: sip:17772939780@callcentric.com
Voice-Message: 0/0 (0/0)
Fax-Message: 0/0 (0/0)
None: 0/0 (0/0)
--- (11 headers 5 lines) ---
Transmitting (NAT) to 204.11.192.22:5060:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP 204.11.192.22:5060;branch=z9hG4bK-d2cce413e7e1e4ed0ec07ce0df0f3667;received=204.11.192.22
From: <sip:17772939780@callcentric.com:5062>
To: <sip:17772939780@callcentric.com>;tag=as0f951acc
Call-ID: 76258d2f8bbc53ea735ef2ebd9a075d6-05f52368f794a3d01c2ea3446313dc66@callcentric.com
CSeq: 1 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
evpp2*CLI>
<-- SIP read from 204.11.192.22:5060:
SIP/2.0 200 Ok
v: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK1985aa19;rport=5060
f: <sip:17772939780@callcentric.com>;tag=as0d2b0762
t: <sip:17772939780@callcentric.com>
i: 6880133c7f613ea67b07b9363907137c@192.168.0.103
CSeq: 1812 REGISTER
m: <sip:s@67.161.185.121>;expires=60
l: 0
--- (8 headers 0 lines) ---
Scheduling destruction of call '6880133c7f613ea67b07b9363907137c@192.168.0.103' in 32000 ms
-- Executing Macro("Zap/6-1", "exten-vm|7240|7240") in new stack
-- Executing Macro("Zap/6-1", "user-callerid") in new stack
-- Executing NoOp("Zap/6-1", "user-callerid: device 233") in new stack
-- Executing GotoIf("Zap/6-1", "0?report") in new stack
-- Executing GotoIf("Zap/6-1", "0?start") in new stack
-- Executing Set("Zap/6-1", "REALCALLERIDNUM=233") in new stack
-- Executing NoOp("Zap/6-1", "REALCALLERIDNUM is 233") in new stack
-- Executing Set("Zap/6-1", "AMPUSER=233") in new stack
-- Executing Set("Zap/6-1", "AMPUSERCIDNAME=Adeel") in new stack
-- Executing GotoIf("Zap/6-1", "0?report") in new stack
-- Executing Set("Zap/6-1", "CALLERID(all)=Adeel <233>") in new stack
-- Executing Set("Zap/6-1", "REALCALLERIDNUM=233") in new stack
-- Executing NoOp("Zap/6-1", "TTL: ARG1: 7240") in new stack
-- Executing GotoIf("Zap/6-1", "0?continue") in new stack
-- Executing Set("Zap/6-1", "__TTL=64") in new stack
-- Executing GotoIf("Zap/6-1", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("Zap/6-1", "Using CallerID "Adeel" <233>") in new stack
-- Executing Set("Zap/6-1", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("Zap/6-1", "VMBOX=7240") in new stack
-- Executing Set("Zap/6-1", "EXTTOCALL=7240") in new stack
-- Executing Set("Zap/6-1", "CFUEXT=") in new stack
-- Executing Set("Zap/6-1", "CFBEXT=") in new stack
-- Executing Set("Zap/6-1", "RT=15") in new stack
-- Executing Macro("Zap/6-1", "record-enable|7240|IN") in new stack
-- Executing GotoIf("Zap/6-1", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("Zap/6-1", "recordingcheck|20070623-181142|1182647495.141") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070623-181142|1182647495.141: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("Zap/6-1", "No recording needed") in new stack
-- Executing Macro("Zap/6-1", "dial|15|trw|7240") in new stack
-- Executing DeadAGI("Zap/6-1", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: priority is 1
dialparties.agi: Caller ID name is 'Adeel' number is '233'
dialparties.agi: Methodology of ring is 'none'
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
-- dialparties.agi: Added extension 7240 to extension map
-- dialparties.agi: Extension 7240 cf is disabled
-- dialparties.agi: Extension 7240 do not disturb is disabled
> dialparties.agi: extnum: 7240
> dialparties.agi: exthascw: 1
> dialparties.agi: exthascfb: 0
> dialparties.agi: extcfb:
> dialparties.agi: exthascfu: 0
> dialparties.agi: extcfu:
-- dialparties.agi: dbset CALLTRACE/7240 to 233
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("Zap/6-1", "SIP/7240|15|trw") in new stack
We're at 192.168.0.103 port 14328
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 16 lines
Reliably Transmitting (NAT) to 192.168.0.104:5060:
INVITE sip:7240@67.161.185.121:32776;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1f5b51c7;rport
From: "Adeel" <sip:233@192.168.0.103>;tag=as08aa317f
To: <sip:7240@67.161.185.121:32776;user=phone>
Contact: <sip:233@192.168.0.103>
Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 24 Jun 2007 01:11:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 365
v=0
=root 5804 5804 IN IP4 192.168.0.103
s=session
c=IN IP4 192.168.0.103
t=0 0
m=audio 14328 RTP/AVP 18 3 110 97 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 7240
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1f5b51c7;rport
From: "Adeel" <sip:233@192.168.0.103>;tag=as08aa317f
To: <sip:7240@67.161.185.121:32776;user=phone>
Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103
CSeq: 102 INVITE
User-Agent: Grandstream BT120 1.0.8.33
Warning: 399 67.161.185.121 "detected NAT type is full cone"
Content-Length: 0
--- (9 headers 0 lines) ---
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1f5b51c7;rport
From: "Adeel" <sip:233@192.168.0.103>;tag=as08aa317f
To: <sip:7240@67.161.185.121:32776;user=phone>;tag=17e302e1cbe45b3d
Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103
CSeq: 102 INVITE
User-Agent: Grandstream BT120 1.0.8.33
Warning: 399 67.161.185.121 "detected NAT type is full cone"
Content-Length: 0
--- (9 headers 0 lines) ---
-- SIP/7240-08218198 is ringing
== Manager 'admin' logged off from 127.0.0.1
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
--- (0 headers 0 lines) Nat keepalive ---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.104:5060:
OPTIONS sip:7240@67.161.185.121:32776;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK5e2fd17b;rport
From: "Unknown" <sip:Unknown@192.168.0.103>;tag=as530907c3
To: <sip:7240@67.161.185.121:32776;user=phone>
Contact: <sip:Unknown@192.168.0.103>
Call-ID: 360f35ea4ff7248a5b1135e72e96488c@192.168.0.103
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 24 Jun 2007 01:11:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK5e2fd17b;rport
From: "Unknown" <sip:Unknown@192.168.0.103>;tag=as530907c3
To: <sip:7240@67.161.185.121:32776;user=phone>;tag=17e302e1cbe45b3d
Call-ID: 360f35ea4ff7248a5b1135e72e96488c@192.168.0.103
CSeq: 102 OPTIONS
User-Agent: Grandstream BT120 1.0.8.33
Warning: 399 67.161.185.121 "detected NAT type is full cone"
Contact: <sip:7240@67.161.185.121:32776;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
--- (12 headers 0 lines) ---
Destroying call '360f35ea4ff7248a5b1135e72e96488c@192.168.0.103'
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1f5b51c7;rport
From: "Adeel" <sip:233@192.168.0.103>;tag=as08aa317f
To: <sip:7240@67.161.185.121:32776;user=phone>;tag=17e302e1cbe45b3d
Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103
CSeq: 102 INVITE
User-Agent: Grandstream BT120 1.0.8.33
Warning: 399 67.161.185.121 "detected NAT type is full cone"
Contact: <sip:7240@67.161.185.121:32776;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 218
v=0
o=7240 8000 8000 IN IP4 67.161.185.121
s=SIP Call
c=IN IP4 67.161.185.121
t=0 0
m=audio 56378 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 67.161.185.121:56378
Found description format G729
Found description format telephone-event
Capabilities: us - 0x70e (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:7240@67.161.185.121:32776;user=phone>
set_destination: Parsing <sip:7240@67.161.185.121:32776;user=phone> for address/port to send to
set_destination: set destination to 67.161.185.121, port 32776
Transmitting (NAT) to 192.168.0.104:5060:
ACK sip:7240@67.161.185.121:32776;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK4e79b957;rport
From: "Adeel" <sip:233@192.168.0.103>;tag=as08aa317f
To: <sip:7240@67.161.185.121:32776;user=phone>;tag=17e302e1cbe45b3d
Contact: <sip:233@192.168.0.103>
Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/7240-08218198 answered Zap/6-1
Scheduling destruction of call '7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103' in 32000 ms
set_destination: Parsing <sip:7240@67.161.185.121:32776;user=phone> for address/port to send to
set_destination: set destination to 67.161.185.121, port 32776
Reliably Transmitting (NAT) to 192.168.0.104:5060:
BYE sip:7240@67.161.185.121:32776;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK7ef46beb;rport
From: "Adeel" <sip:233@192.168.0.103>;tag=as08aa317f
To: <sip:7240@67.161.185.121:32776;user=phone>;tag=17e302e1cbe45b3d
Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/6-1' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/6-1' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/6-1'
-- Executing Macro("Zap/6-1", "hangupcall") in new stack
-- Executing ResetCDR("Zap/6-1", "w") in new stack
-- Executing NoCDR("Zap/6-1", "") in new stack
-- Executing GotoIf("Zap/6-1", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("Zap/6-1", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("Zap/6-1", "5") in new stack
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK7ef46beb;rport
From: "Adeel" <sip:233@192.168.0.103>;tag=as08aa317f
To: <sip:7240@67.161.185.121:32776;user=phone>;tag=17e302e1cbe45b3d
Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103
CSeq: 103 BYE
User-Agent: Grandstream BT120 1.0.8.33
Warning: 399 67.161.185.121 "detected NAT type is full cone"
Contact: <sip:7240@67.161.185.121:32776;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
--- (12 headers 0 lines) ---
Destroying call '7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103'
-- Executing Hangup("Zap/6-1", "") in new stack
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'Zap/6-1' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'Zap/6-1'
-- Hungup 'Zap/6-1'
Destroying call '51274156501c534a5d9ac4f042b62766@192.168.0.103'
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.34.176.212:5060:
OPTIONS sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK2da9b019;rport
From: "Unknown" <sip:Unknown@67.161.185.121>;tag=as31184c0c
To: <sip:did.voip.les.net>
Contact: <sip:Unknown@67.161.185.121>
Call-ID: 642376b9719e61d6212f844015cc0ab1@67.161.185.121
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 24 Jun 2007 01:11:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
evpp2*CLI>
<-- SIP read from 64.34.176.212:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK2da9b019;received=67.161.185.121;rport=5060
From: "Unknown" <sip:Unknown@67.161.185.121>;tag=as31184c0c
To: <sip:did.voip.les.net>;tag=as55956209
Call-ID: 642376b9719e61d6212f844015cc0ab1@67.161.185.121
CSeq: 102 OPTIONS
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0
--- (10 headers 0 lines) ---
Destroying call '642376b9719e61d6212f844015cc0ab1@67.161.185.121'
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
INVITE sip:233@192.168.0.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK6479bd648933e19c
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>
Contact: <sip:7240@67.161.185.121:32776;user=phone>
Supported: replaces
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49459 INVITE
User-Agent: Grandstream BT120 1.0.8.33
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 387
v=0
o=7240 8000 8000 IN IP4 67.161.185.121
s=SIP Call
c=IN IP4 67.161.185.121
t=0 0
m=audio 56378 RTP/AVP 0 8 4 18 2 97 9 101
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=20
a=rtpmap:9 G722/16000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 18 lines) ---
Using INVITE request as basis request - 48adb4a1972878bc@192.168.0.104
Sending to 67.161.185.121 : 32776 (NAT)
Reliably Transmitting (NAT) to 192.168.0.104:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK6479bd648933e19c;received=192.168.0.104
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>;tag=as44dd4976
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49459 INVITE
User-Agent: Asterisk PBX
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="57b0c117"
Content-Length: 0
---
Scheduling destruction of call '48adb4a1972878bc@192.168.0.104' in 15000 ms
Found user '7240'
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
ACK sip:233@192.168.0.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK6479bd648933e19c
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>;tag=as44dd4976
Contact: <sip:7240@67.161.185.121:32776;user=phone>
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49459 ACK
User-Agent: Grandstream BT120 1.0.8.33
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (11 headers 0 lines) ---
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
INVITE sip:233@192.168.0.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bKebe3ef55f1e35855
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>
Contact: <sip:7240@67.161.185.121:32776;user=phone>
Supported: replaces
Proxy-Authorization: Digest username="7240", realm="asterisk", algorithm=MD5, uri="sip:233@192.168.0.103;user=phone", nonce="57b0c117", response="6cc9c5fdc8366b049d225aae4ef54451"
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49460 INVITE
User-Agent: Grandstream BT120 1.0.8.33
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 387
v=0
o=7240 8000 8001 IN IP4 67.161.185.121
s=SIP Call
c=IN IP4 67.161.185.121
t=0 0
m=audio 56378 RTP/AVP 0 8 4 18 2 97 9 101
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=20
a=rtpmap:9 G722/16000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (14 headers 18 lines) ---
Using INVITE request as basis request - 48adb4a1972878bc@192.168.0.104
Sending to 67.161.185.121 : 32776 (NAT)
Found user '7240'
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 101
Peer audio RTP is at port 67.161.185.121:56378
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format iLBC
Found description format G722
Found description format telephone-event
Capabilities: us - 0x70e (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x50c (ulaw|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 233 in from-internal (domain 192.168.0.103;user=phone)
list_route: hop: <sip:7240@67.161.185.121:32776;user=phone>
Transmitting (NAT) to 192.168.0.104:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bKebe3ef55f1e35855;received=192.168.0.104
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49460 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:233@192.168.0.103>
Content-Length: 0
---
-- Executing Macro("SIP/7240-08201ed0", "exten-vm|233|233") in new stack
-- Executing Macro("SIP/7240-08201ed0", "user-callerid") in new stack
-- Executing NoOp("SIP/7240-08201ed0", "user-callerid: device 7240") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?report") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?start") in new stack
-- Executing Set("SIP/7240-08201ed0", "REALCALLERIDNUM=7240") in new stack
-- Executing NoOp("SIP/7240-08201ed0", "REALCALLERIDNUM is 7240") in new stack
-- Executing Set("SIP/7240-08201ed0", "AMPUSER=7240") in new stack
-- Executing Set("SIP/7240-08201ed0", "AMPUSERCIDNAME=Nancy") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?report") in new stack
-- Executing Set("SIP/7240-08201ed0", "CALLERID(all)=Nancy <7240>") in new stack
-- Executing Set("SIP/7240-08201ed0", "REALCALLERIDNUM=7240") in new stack
-- Executing NoOp("SIP/7240-08201ed0", "TTL: ARG1: 233") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?continue") in new stack
-- Executing Set("SIP/7240-08201ed0", "__TTL=64") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/7240-08201ed0", "Using CallerID "Nancy" <7240>") in new stack
-- Executing Set("SIP/7240-08201ed0", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("SIP/7240-08201ed0", "VMBOX=233") in new stack
-- Executing Set("SIP/7240-08201ed0", "EXTTOCALL=233") in new stack
-- Executing Set("SIP/7240-08201ed0", "CFUEXT=") in new stack
-- Executing Set("SIP/7240-08201ed0", "CFBEXT=") in new stack
-- Executing Set("SIP/7240-08201ed0", "RT=15") in new stack
-- Executing Macro("SIP/7240-08201ed0", "record-enable|233|IN") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/7240-08201ed0", "recordingcheck|20070623-181203|1182647523.143") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
<-- SIP read from 192.168.0.104:5060:
--- (0 headers 0 lines) Nat keepalive ---
recordingcheck|20070623-181203|1182647523.143: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/7240-08201ed0", "No recording needed") in new stack
-- Executing Macro("SIP/7240-08201ed0", "dial|15|trw|233") in new stack
-- Executing DeadAGI("SIP/7240-08201ed0", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: priority is 1
dialparties.agi: Caller ID name is 'Nancy' number is '7240'
dialparties.agi: Methodology of ring is 'none'
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
-- dialparties.agi: Added extension 233 to extension map
-- dialparties.agi: Extension 233 cf is disabled
dialparties.agi: Extension 233 has do not disturb enabled, or followme pre-ring returned busy
-- AGI Script dialparties.agi completed, returning 0
-- Executing NoOp("SIP/7240-08201ed0", "Returned from dialparties with no extensions to call") in new stack
-- Executing NoOp("SIP/7240-08201ed0", "DIALSTATUS is BUSY") in new stack
-- Executing Set("SIP/7240-08201ed0", "SV_DIALSTATUS=BUSY") in new stack
-- Executing GosubIf("SIP/7240-08201ed0", "0?docfu|1") in new stack
-- Executing GosubIf("SIP/7240-08201ed0", "0?docfb|1") in new stack
-- Executing Set("SIP/7240-08201ed0", "DIALSTATUS=BUSY") in new stack
-- Executing NoOp("SIP/7240-08201ed0", "Voicemail is 233") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?s-BUSY|1") in new stack
-- Executing NoOp("SIP/7240-08201ed0", "Sending to Voicemail box 233") in new stack
-- Executing Macro("SIP/7240-08201ed0", "vm|233|BUSY") in new stack
-- Executing Macro("SIP/7240-08201ed0", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/7240-08201ed0", "user-callerid: Nancy 7240") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?report") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "1?start") in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp("SIP/7240-08201ed0", "REALCALLERIDNUM is 7240") in new stack
-- Executing Set("SIP/7240-08201ed0", "AMPUSER=7240") in new stack
-- Executing Set("SIP/7240-08201ed0", "AMPUSERCIDNAME=Nancy") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?report") in new stack
-- Executing Set("SIP/7240-08201ed0", "CALLERID(all)=Nancy <7240>") in new stack
-- Executing Set("SIP/7240-08201ed0", "REALCALLERIDNUM=7240") in new stack
-- Executing NoOp("SIP/7240-08201ed0", "TTL: 64 ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/7240-08201ed0", "Using CallerID "Nancy" <7240>") in new stack
-- Executing Set("SIP/7240-08201ed0", "VMGAIN=") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "1?s-BUSY|1") in new stack
-- Goto (macro-vm,s-BUSY,1)
-- Executing NoOp("SIP/7240-08201ed0", "BUSY voicemail") in new stack
-- Executing Macro("SIP/7240-08201ed0", "get-vmcontext|233") in new stack
-- Executing Set("SIP/7240-08201ed0", "VMCONTEXT=default") in new stack
-- Executing GotoIf("SIP/7240-08201ed0", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing NoOp("SIP/7240-08201ed0", "") in new stack
-- Executing VoiceMail("SIP/7240-08201ed0", "233@default|b") in new stack
We're at 192.168.0.103 port 16160
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.104:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bKebe3ef55f1e35855;received=192.168.0.104
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>;tag=as1feab7b9
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49460 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:233@192.168.0.103>
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 5804 5804 IN IP4 192.168.0.103
s=session
c=IN IP4 192.168.0.103
t=0 0
m=audio 16160 RTP/AVP 18 97 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/7240-08201ed0' in macro 'vm'
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/7240-08201ed0' in macro 'exten-vm'
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/7240-08201ed0'
Scheduling destruction of call '48adb4a1972878bc@192.168.0.104' in 32000 ms
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
ACK sip:233@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK8d746d15b4a57536
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>;tag=as1feab7b9
Contact: <sip:7240@67.161.185.121:32776;user=phone>
Proxy-Authorization: Digest username="7240", realm="asterisk", algorithm=MD5, uri="sip:233@192.168.0.103", nonce="57b0c117", response="69bcdce490eb32d9a650d4548ab2955e"
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49460 ACK
User-Agent: Grandstream BT120 1.0.8.33
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:7240@67.161.185.121:32776;user=phone> for address/port to send to
set_destination: set destination to 67.161.185.121, port 32776
Reliably Transmitting (NAT) to 192.168.0.104:5060:
CANCEL :233@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK662f2add;rport
From: <sip:233@192.168.0.103;user=phone>
To: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call '48adb4a1972878bc@192.168.0.104' in 32000 ms
== Manager 'admin' logged off from 127.0.0.1
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
SIP/2.0 481 No Such Call
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK662f2add;rport
From: <sip:233@192.168.0.103;user=phone>
To: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 101 CANCEL
User-Agent: Grandstream BT120 1.0.8.33
Warning: 399 67.161.185.121 "detected NAT type is full cone"
Content-Length: 0
--- (9 headers 0 lines) ---
12 headers, 0 lines
Reliably Transmitting (NAT) to 204.147.183.18:5060:
OPTIONS sip:sip.stanaphone.com SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK66820565;rport
From: "Unknown" <sip:Unknown@67.161.185.121>;tag=as21a81c99
To: <sip:sip.stanaphone.com>
Contact: <sip:Unknown@67.161.185.121>
Call-ID: 6026db9e78e91c9371f868b403589171@67.161.185.121
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 24 Jun 2007 01:12:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
evpp2*CLI>
<-- SIP read from 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK66820565;rport=5060
From: "Unknown" <sip:Unknown@67.161.185.121>;tag=as21a81c99
To: <sip:sip.stanaphone.com>;tag=4e487ad2f2fae333606267674b3a79b1.64d6
Call-ID: 6026db9e78e91c9371f868b403589171@67.161.185.121
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.9.5-pre1 (i386/linux))
Content-Length: 0
Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=29582 req_src_ip=67.161.185.121 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1"
--- (9 headers 0 lines) ---
Destroying call '6026db9e78e91c9371f868b403589171@67.161.185.121'
Destroying call '73bb50220272cd587bc9342a325f9afd@192.168.0.103'
Destroying call '6880133c7f613ea67b07b9363907137c@192.168.0.103'
evpp2*CLI>
<-- SIP read from 192.168.0.104:5060:
BYE sip:233@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK420487a531a54536
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>;tag=as1feab7b9
Supported: replaces
Proxy-Authorization: Digest username="7240", realm="asterisk", algorithm=MD5, uri="sip:233@192.168.0.103", nonce="57b0c117", response="be4e5b317dfdde22c97aa6b588057239"
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49461 BYE
User-Agent: Grandstream BT120 1.0.8.33
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (12 headers 0 lines) ---
Sending to 67.161.185.121 : 32776 (NAT)
Transmitting (NAT) to 192.168.0.104:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK420487a531a54536;received=192.168.0.104
From: "Nancy Spicer" <sip:7240@192.168.0.103;user=phone>;tag=f9eba7edd6dc2d56
To: <sip:233@192.168.0.103;user=phone>;tag=as1feab7b9
Call-ID: 48adb4a1972878bc@192.168.0.104
CSeq: 49461 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:233@192.168.0.103>
Content-Length: 0
---
Destroying call '48adb4a1972878bc@192.168.0.104'