[root@pbx01 ~]# asterisk -r
Asterisk 1.8.9.3, 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.
=========================================================================
Connected to Asterisk 1.8.9.3 currently running on pbx01 (pid = 2906)
Verbosity is at least 1000000

<--- SIP read from UDP:192.168.10.254:5060 --->
INVITE sip:01299252388@192.168.10.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-3f457846
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Test Extension" <sip:777@192.168.10.254:5060>
Expires: 240
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 312
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 360949 360949 IN IP4 192.168.10.254
s=-
c=IN IP4 192.168.10.254
t=0 0
m=audio 16386 RTP/AVP 0 8 2 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.10.254:5060 (NAT)
Using INVITE request as basis request - 8ea9fae6-b8386c6@192.168.80.3
Found peer '777' for '777' from 192.168.10.254:5060

<--- Reliably Transmitting (no NAT) to 192.168.10.254:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-3f457846;received=192.168.10.254
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>;tag=as0dafa630
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 101 INVITE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06c3ea5d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8ea9fae6-b8386c6@192.168.80.3' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.10.254:5060 --->
ACK sip:01299252388@192.168.10.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-3f457846
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>;tag=as0dafa630
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Test Extension" <sip:777@192.168.10.254:5060>
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.10.254:5060 --->
INVITE sip:01299252388@192.168.10.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-b8c4204e
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="777",realm="asterisk",nonce="06c3ea5d",uri="sip:01299252388@192.168.10.200",algorithm=MD5,response="9407bd1a3a1b6af5bfc4818d2b2754cf"
Contact: "Test Extension" <sip:777@192.168.10.254:5060>
Expires: 240
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 312
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 360949 360949 IN IP4 192.168.10.254
s=-
c=IN IP4 192.168.10.254
t=0 0
m=audio 16386 RTP/AVP 0 8 2 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.10.254:5060 (no NAT)
Using INVITE request as basis request - 8ea9fae6-b8386c6@192.168.80.3
Found peer '777' for '777' from 192.168.10.254:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 9
Found RTP audio format 18
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 G726-32 for ID 2
Found audio description format G722 for ID 9
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x190c (ulaw|alaw|g726|g729|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.10.254:16386
Looking for 01299252388 in from-internal (domain 192.168.10.200)
list_route: hop: <sip:777@192.168.10.254:5060>

<--- Transmitting (no NAT) to 192.168.10.254:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-b8c4204e;received=192.168.10.254
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 102 INVITE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:01299252388@192.168.10.200:5060>
Content-Length: 0


<------------>
    -- Executing [01299252388@from-internal:1] Macro("SIP/777-000000e0", "user-callerid,LIMIT,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/777-000000e0", "AMPUSER=777") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/777-000000e0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/777-000000e0", "1?Set(REALCALLERIDNUM=777)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/777-000000e0", "AMPUSER=777") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/777-000000e0", "AMPUSERCIDNAME=Chris Pottrell") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/777-000000e0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/777-000000e0", "AMPUSERCID=777") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/777-000000e0", "CALLERID(all)="Chris Pottrell" <777>") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/777-000000e0", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/777-000000e0", "1?Set(GROUP(concurrency_limit)=777)") in new stack
    -- Executing [s@macro-user-callerid:11] ExecIf("SIP/777-000000e0", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:12] GosubIf("SIP/777-000000e0", "7?sub-ccss,s,1(from-internal,01299252388)") in new stack
    -- Executing [s@sub-ccss:1] ExecIf("SIP/777-000000e0", "0?Return()") in new stack
    -- Executing [s@sub-ccss:2] Set("SIP/777-000000e0", "CCSS_SETUP=TRUE") in new stack
    -- Executing [s@sub-ccss:3] GosubIf("SIP/777-000000e0", "0?monitor_config,1(from-internal,01299252388):monitor_default,1(from-internal,01299252388)") in new stack
    -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/777-000000e0", "0?is_exten") in new stack
    -- Executing [monitor_default@sub-ccss:2] StackPop("SIP/777-000000e0", "") in new stack
    -- Executing [monitor_default@sub-ccss:3] Return("SIP/777-000000e0", "FALSE") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/777-000000e0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,26)
    -- Executing [s@macro-user-callerid:26] Set("SIP/777-000000e0", "CALLERID(number)=777") in new stack
    -- Executing [s@macro-user-callerid:27] Set("SIP/777-000000e0", "CALLERID(name)=Chris Pottrell") in new stack
    -- Executing [s@macro-user-callerid:28] Set("SIP/777-000000e0", "CHANNEL(language)=en") in new stack
    -- Executing [01299252388@from-internal:2] Set("SIP/777-000000e0", "MOHCLASS=default") in new stack
    -- Executing [01299252388@from-internal:3] Set("SIP/777-000000e0", "_NODEST=") in new stack
    -- Executing [01299252388@from-internal:4] Gosub("SIP/777-000000e0", "sub-record-check,s,1(out,01299252388,)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/777-000000e0", "1?check") in new stack
    -- Goto (sub-record-check,s,3)
    -- Executing [s@sub-record-check:3] Set("SIP/777-000000e0", "MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:4] GotoIf("SIP/777-000000e0", "1?next") in new stack
    -- Goto (sub-record-check,s,7)
    -- Executing [s@sub-record-check:7] ExecIf("SIP/777-000000e0", "0?Return()") in new stack
    -- Executing [s@sub-record-check:8] GotoIf("SIP/777-000000e0", "0?out,1") in new stack
    -- Executing [s@sub-record-check:9] Set("SIP/777-000000e0", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:10] ExecIf("SIP/777-000000e0", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [s@sub-record-check:11] Set("SIP/777-000000e0", "NOW=1331128924") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/777-000000e0", "__DAY=07") in new stack
    -- Executing [s@sub-record-check:13] Set("SIP/777-000000e0", "__MONTH=03") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/777-000000e0", "__YEAR=2012") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/777-000000e0", "__TIMESTR=20120307-140204") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/777-000000e0", "__FROMEXTEN=777") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/777-000000e0", "__CALLFILENAME=out-01299252388-777-20120307-140204-1331128924.224") in new stack
    -- Executing [s@sub-record-check:18] Goto("SIP/777-000000e0", "out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] ExecIf("SIP/777-000000e0", "1?Set(__REC_POLICY_MODE=always)") in new stack
    -- Executing [out@sub-record-check:2] GosubIf("SIP/777-000000e0", "1?record,1(exten,01299252388,777)") in new stack
    -- Executing [record@sub-record-check:1] Set("SIP/777-000000e0", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
    -- Executing [record@sub-record-check:2] MixMonitor("SIP/777-000000e0", "2012/03/07/out-01299252388-777-20120307-140204-1331128924.224.wav,,") in new stack
    -- Executing [record@sub-record-check:3] Set("SIP/777-000000e0", "__REC_STATUS=RECORDING") in new stack
    -- Executing [record@sub-record-check:4] Set("SIP/777-000000e0", "CDR(recordingfile)=out-01299252388-777-20120307-140204-1331128924.224.wav") in new stack
    -- Executing [record@sub-record-check:5] Return("SIP/777-000000e0", "") in new stack
    -- Executing [out@sub-record-check:3] Return("SIP/777-000000e0", "") in new stack
    -- Executing [01299252388@from-internal:5] Macro("SIP/777-000000e0", "dialout-trunk,2,01299252388,") in new stack
  == Begin MixMonitor Recording SIP/777-000000e0
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/777-000000e0", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/777-000000e0", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/777-000000e0", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/777-000000e0", "DIAL_NUMBER=01299252388") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/777-000000e0", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/777-000000e0", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/777-000000e0", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/777-000000e0", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/777-000000e0", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/777-000000e0", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/777-000000e0", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/777-000000e0", "0?Set(REALCALLERIDNUM=777)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/777-000000e0", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/777-000000e0", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/777-000000e0", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/777-000000e0", "TRUNKOUTCID=01179113714") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/777-000000e0", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/777-000000e0", "1?Set(CALLERID(all)=01179113714)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/777-000000e0", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/777-000000e0", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/777-000000e0", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/777-000000e0", "0?sub-flp-2,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/777-000000e0", "OUTNUM=01299252388") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/777-000000e0", "custom=SIP/VoipTalk") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/777-000000e0", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/777-000000e0", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/777-000000e0", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/777-000000e0", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/777-000000e0", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/777-000000e0", "1?Set(CONNECTEDLINE(num,i)=01299252388)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/777-000000e0", "1?Set(CONNECTEDLINE(name,i)=CID:01179113714)") in new stack
    -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/777-000000e0", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/777-000000e0", "SIP/VoipTalk/01299252388,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
We think we can do text
Audio is at 18580
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 77.240.48.94:5060:
INVITE sip:01299252388@voiptalk.org SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK46c5e422
Max-Forwards: 70
From: "01179113714" <sip:01179113714@212.74.46.35>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>
Contact: <sip:01179113714@212.74.46.35:5060>
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 14:02:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 521

v=0
o=root 1890988409 1890988409 IN IP4 212.74.46.35
s=Asterisk PBX 1.8.9.3
c=IN IP4 212.74.46.35
t=0 0
m=audio 18580 RTP/AVP 8 0 3 112 5 10 7 18 111 9 118 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/VoipTalk/01299252388

<--- SIP read from UDP:77.240.48.94:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK46c5e422
From: "01179113714" <sip:01179113714@192.168.10.200:5060>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 102 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:77.240.48.94:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK46c5e422
From: "01179113714" <sip:01179113714@192.168.10.200:5060>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>;tag=fd79486175647ed1617969929fdaad02.6e58
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="voiptalk.org", nonce="4f576a82000032ecfdb89c2272d4432a1c61f29065f1a6d1"
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:01299252388@voiptalk.org> for address/port to send to
set_destination: set destination to 77.240.48.94:5060
Transmitting (no NAT) to 77.240.48.94:5060:
ACK sip:01299252388@voiptalk.org SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK46c5e422
Max-Forwards: 70
From: "01179113714" <sip:01179113714@212.74.46.35>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>;tag=fd79486175647ed1617969929fdaad02.6e58
Contact: <sip:01179113714@212.74.46.35:5060>
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.0(1.8.9.3)
Content-Length: 0


---
We think we can do text
Audio is at 18580
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 77.240.48.94:5060:
INVITE sip:01299252388@voiptalk.org SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK6275c0a6
Max-Forwards: 70
From: "01179113714" <sip:01179113714@212.74.46.35>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>
Contact: <sip:01179113714@212.74.46.35:5060>
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.10.0(1.8.9.3)
Proxy-Authorization: Digest username="844238829", realm="voiptalk.org", algorithm=MD5, uri="sip:01299252388@voiptalk.org", nonce="4f576a82000032ecfdb89c2272d4432a1c61f29065f1a6d1", response="ef25b1cc156062cccfba4ce9aa499d03"
Date: Wed, 07 Mar 2012 14:02:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 521

v=0
o=root 1890988409 1890988410 IN IP4 212.74.46.35
s=Asterisk PBX 1.8.9.3
c=IN IP4 212.74.46.35
t=0 0
m=audio 18580 RTP/AVP 8 0 3 112 5 10 7 18 111 9 118 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:77.240.48.94:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK6275c0a6
From: "01179113714" <sip:01179113714@192.168.10.200:5060>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 103 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:77.240.48.94:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK6275c0a6
From: "01179113714" <sip:01179113714@192.168.10.200:5060>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 103 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:77.240.48.94:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK6275c0a6
Record-Route: <sip:77.240.48.94;lr=on;ftag=as4ffc9dcc>
From: "01179113714" <sip:01179113714@192.168.10.200:5060>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>;tag=as066182b7
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 103 INVITE
Server: voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:CALL-70692040-01299252388@77.240.54.11>
Content-Type: application/sdp
Content-Length: 339

v=0
o=voip 936727554 936727554 IN IP4 77.240.54.11
s=voip
c=IN IP4 77.240.54.11
t=0 0
m=audio 15120 RTP/AVP 8 0 3 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 16 lines) ---
list_route: hop: <sip:77.240.48.94;lr=on;ftag=as4ffc9dcc>
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 77.240.54.11:15120
    -- SIP/VoipTalk-000000e1 is making progress passing it to SIP/777-000000e0
Audio is at 10444
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.10.254:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-b8c4204e;received=192.168.10.254
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>;tag=as7cedfa2c
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 102 INVITE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:01299252388@192.168.10.200:5060>
Content-Type: application/sdp
Content-Length: 362

v=0
o=root 1135164252 1135164252 IN IP4 192.168.10.200
s=Asterisk PBX 1.8.9.3
c=IN IP4 192.168.10.200
t=0 0
m=audio 10444 RTP/AVP 8 0 18 2 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:77.240.48.94:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK6275c0a6
Record-Route: <sip:77.240.48.94;lr=on;ftag=as4ffc9dcc>
From: "01179113714" <sip:01179113714@192.168.10.200:5060>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>;tag=as066182b7
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 103 INVITE
Server: voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:CALL-70692040-01299252388@77.240.54.11>
Content-Type: application/sdp
Content-Length: 339

v=0
o=voip 936727554 936727555 IN IP4 77.240.54.11
s=voip
c=IN IP4 77.240.54.11
t=0 0
m=audio 15120 RTP/AVP 8 0 3 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 16 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 77.240.54.11:15120
list_route: hop: <sip:77.240.48.94;lr=on;ftag=as4ffc9dcc>
set_destination: Parsing <sip:77.240.48.94;lr=on;ftag=as4ffc9dcc> for address/port to send to
set_destination: set destination to 77.240.48.94:5060
Transmitting (no NAT) to 77.240.48.94:5060:
ACK sip:CALL-70692040-01299252388@77.240.54.11 SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK1c914af8
Route: <sip:77.240.48.94;lr=on;ftag=as4ffc9dcc>
Max-Forwards: 70
From: "01179113714" <sip:01179113714@212.74.46.35>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>;tag=as066182b7
Contact: <sip:01179113714@212.74.46.35:5060>
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 103 ACK
User-Agent: FPBX-2.10.0(1.8.9.3)
Content-Length: 0


---
    -- SIP/VoipTalk-000000e1 answered SIP/777-000000e0
Audio is at 10444
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.10.254:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-b8c4204e;received=192.168.10.254
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>;tag=as7cedfa2c
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 102 INVITE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:01299252388@192.168.10.200:5060>
Content-Type: application/sdp
Content-Length: 362

v=0
o=root 1135164252 1135164253 IN IP4 192.168.10.200
s=Asterisk PBX 1.8.9.3
c=IN IP4 192.168.10.200
t=0 0
m=audio 10444 RTP/AVP 8 0 18 2 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.10.254:5060 --->
ACK sip:01299252388@192.168.10.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-29a2dbfd
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>;tag=as7cedfa2c
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="777",realm="asterisk",nonce="06c3ea5d",uri="sip:01299252388@192.168.10.200",algorithm=MD5,response="9407bd1a3a1b6af5bfc4818d2b2754cf"
Contact: "Test Extension" <sip:777@192.168.10.254:5060>
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.10.254:1031:
OPTIONS sip:101@192.168.10.254:1031 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4450645b
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.10.200>;tag=as7b5b1e1f
To: <sip:101@192.168.10.254:1031>
Contact: <sip:Unknown@192.168.10.200:5060>
Call-ID: 4993fea9746ff3754247271c3d0900e5@192.168.10.200:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 14:02:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (no NAT) to 192.168.10.254:1030:
OPTIONS sip:111@192.168.10.254:1030 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK3ebb2505
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.10.200>;tag=as554d60a4
To: <sip:111@192.168.10.254:1030>
Contact: <sip:Unknown@192.168.10.200:5060>
Call-ID: 6bab33477b02a5185471264f6c688c3b@192.168.10.200:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 14:02:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (no NAT) to 192.168.10.254:5060:
OPTIONS sip:777@192.168.10.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK764ec616
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.10.200>;tag=as09f28b2e
To: <sip:777@192.168.10.254:5060>
Contact: <sip:Unknown@192.168.10.200:5060>
Call-ID: 0238f21e0f8d7e205bbf06c4601d3895@192.168.10.200:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 14:02:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.10.254:1031 --->
SIP/2.0 200 OK
To: <sip:101@192.168.10.254:1031>;tag=e2e2004333f3fff1i0
From: "Unknown" <sip:Unknown@192.168.10.200>;tag=as7b5b1e1f
Call-ID: 4993fea9746ff3754247271c3d0900e5@192.168.10.200:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4450645b
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4993fea9746ff3754247271c3d0900e5@192.168.10.200:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.10.254:1030 --->
SIP/2.0 200 OK
To: <sip:111@192.168.10.254:1030>;tag=76576fa3f63e57a9i0
From: "Unknown" <sip:Unknown@192.168.10.200>;tag=as554d60a4
Call-ID: 6bab33477b02a5185471264f6c688c3b@192.168.10.200:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK3ebb2505
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6bab33477b02a5185471264f6c688c3b@192.168.10.200:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.10.254:5060 --->
SIP/2.0 200 OK
To: <sip:777@192.168.10.254:5060>;tag=f64e6d8f7e1b8e2ei0
From: "Unknown" <sip:Unknown@192.168.10.200>;tag=as09f28b2e
Call-ID: 0238f21e0f8d7e205bbf06c4601d3895@192.168.10.200:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK764ec616
Server: Cisco/SPA525G2-7.4.9c
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0238f21e0f8d7e205bbf06c4601d3895@192.168.10.200:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.10.254:5060 --->
BYE sip:01299252388@192.168.10.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-11e2c49b
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>;tag=as7cedfa2c
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username="777",realm="asterisk",nonce="06c3ea5d",uri="sip:01299252388@192.168.10.200:5060",algorithm=MD5,response="5f03fd9dc78675446d776889063eebbb"
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.10.254:5060 (no NAT)
Scheduling destruction of SIP dialog '8ea9fae6-b8386c6@192.168.80.3' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.10.254:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-11e2c49b;received=192.168.10.254
From: "Test Extension" <sip:777@192.168.10.200>;tag=ee246c66362fc346o0
To: "01299252388" <sip:01299252388@192.168.10.200>;tag=as7cedfa2c
Call-ID: 8ea9fae6-b8386c6@192.168.80.3
CSeq: 103 BYE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/777-000000e0", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/777-000000e0", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] Hangup("SIP/777-000000e0", "") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/777-000000e0' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/777-000000e0'
Scheduling destruction of SIP dialog '5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:77.240.48.94;lr=on;ftag=as4ffc9dcc> for address/port to send to
set_destination: set destination to 77.240.48.94:5060
Reliably Transmitting (no NAT) to 77.240.48.94:5060:
BYE sip:CALL-70692040-01299252388@77.240.54.11 SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK3dedbfc4
Route: <sip:77.240.48.94;lr=on;ftag=as4ffc9dcc>
Max-Forwards: 70
From: "01179113714" <sip:01179113714@212.74.46.35>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>;tag=as066182b7
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 104 BYE
User-Agent: FPBX-2.10.0(1.8.9.3)
Proxy-Authorization: Digest username="844238829", realm="voiptalk.org", algorithm=MD5, uri="sip:CALL-70692040-01299252388@77.240.54.11", nonce="4f576a82000032ecfdb89c2272d4432a1c61f29065f1a6d1", response="4cad1fc64a733785908fbd07d94aedbc"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/777-000000e0' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 01299252388, 5) exited non-zero on 'SIP/777-000000e0'
  == MixMonitor close filestream
  == End MixMonitor Recording SIP/777-000000e0

<--- SIP read from UDP:77.240.48.94:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK3dedbfc4
From: "01179113714" <sip:01179113714@192.168.10.200:5060>;tag=as4ffc9dcc
To: <sip:01299252388@voiptalk.org>;tag=as066182b7
Call-ID: 5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060
CSeq: 104 BYE
Server: voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5921b27f11503d022a0ed6dd7928ac80@212.74.46.35:5060' Method: INVITE
pbx01*CLI>
Disconnected from Asterisk server