ASTERISK BACK TO BACK

##########################################################################################
1.1.1.1 = The IP Handset initiating the call (Original ip was the WAN IP of the office )
192.168.10.1 = This IP the internal (LAN) ip of the IP handset

77.77.77.66 = This is the IP of the A2B server that the handset is directly connnected to.
77.77.77.88 = This is the IP of the second A2B server that is supposed to bill the calls made by the first server.

A2B-SECOND-SERVER  = Trunk Name to the second server.

441234567890 is the Tel number called
##########################################################################################


LOG


Asterisk 1.6.2.24, Copyright (C) 1999 - 2010 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.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
[0;37m[1;30m  == [0mParsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf
[1;30m  == [0mFound
[0mConnected to Asterisk 1.6.2.24 currently running on server066 (pid = 6807)
server066*CLI>
[0KVerbosity is at least 3
Core debug is at least 3


[Kserver066*CLI> server066*CLI> server066*CLI>
[0K
<--- SIP read from UDP:1.1.1.1:53450 --->
INVITE sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1029059428From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>Call-ID: 1953639537@192.168.10.1CSeq: 1 INVITEContact: <sip:0024476691@192.168.10.1:5063>Content-Type: application/sdpAllow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGEMax-Forwards: 70User-Agent: Yealink SIP-T26P 6.61.0.83Supported: replacesAllow-Events: talk,hold,conference,refer,check-syncContent-Length: 305v=0o=- 20295 20295 IN IP4 192.168.10.1s=SDP datac=IN IP4 192.168.10.1t=0 0m=audio 4016 RTP/AVP 0 8 9 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:9 G722/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=fmtp:101 0-15a=rtpmap:101 telephone-event/8000a=ptime:20a=sendrecv
<------------->
--- (14 headers 15 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
Sending to 192.168.10.1 : 5063 (no NAT)
Using INVITE request as basis request - 1953639537@192.168.10.1
Found peer '0024476691' for '0024476691' from 1.1.1.1:53450

<--- Reliably Transmitting (NAT) to 1.1.1.1:53450 --->
SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1029059428;received=1.1.1.1From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>;tag=as13a779f6Call-ID: 1953639537@192.168.10.1CSeq: 1 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d912090"Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1953639537@192.168.10.1' in 32000 ms (Method: INVITE)

[Kserver066*CLI>
[0K
<--- SIP read from UDP:1.1.1.1:53450 --->
ACK sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1029059428From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>;tag=as13a779f6Call-ID: 1953639537@192.168.10.1CSeq: 1 ACKContent-Length: 0
<------------->
--- (7 headers 0 lines) ---

[Kserver066*CLI>
[0K
<--- SIP read from UDP:1.1.1.1:53450 --->
INVITE sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1836708136From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>Call-ID: 1953639537@192.168.10.1CSeq: 2 INVITEContact: <sip:0024476691@192.168.10.1:5063>Authorization: Digest username="0024476691", realm="asterisk", nonce="7d912090", uri="sip:441234567890@77.77.77.66", response="d068bcf2769ef7b3699fee63538dc300", algorithm=MD5Content-Type: application/sdpAllow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGEMax-Forwards: 70User-Agent: Yealink SIP-T26P 6.61.0.83Supported: replacesAllow-Events: talk,hold,conference,refer,check-syncContent-Length: 305v=0o=- 20295 20295 IN IP4 192.168.10.1s=SDP datac=IN IP4 192.168.10.1t=0 0m=audio 4016 RTP/AVP 0 8 9 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:9 G722/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=fmtp:101 0-15a=rtpmap:101 telephone-event/8000a=ptime:20a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 1.1.1.1 : 53450 (NAT)
Using INVITE request as basis request - 1953639537@192.168.10.1
Found peer '0024476691' for '0024476691' from 1.1.1.1:53450
Found RTP audio format 0
Found RTP audio format 8
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 G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (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 192.168.10.1:4016
Looking for 441234567890 in a2billing (domain 77.77.77.66)
list_route: hop: <sip:0024476691@192.168.10.1:5063>

<--- Transmitting (NAT) to 1.1.1.1:53450 --->
SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1836708136;received=1.1.1.1From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>Call-ID: 1953639537@192.168.10.1CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.66>Content-Length: 0
<------------>

[Kserver066*CLI>
[0K    -- Executing [441234567890@a2billing:1] [1;36mAGI[0m("[1;35mSIP/0024476691-00000004[0m", "[1;35ma2billing.php,1[0m") in new stack

[Kserver066*CLI>
[0K    -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php

[Kserver066*CLI>
[0K    -- AGI Script Executing Application: (DIAL) Options: (SIP/A2B-SECOND-SERVER/441234567890,60,S(5088000:))

[Kserver066*CLI>
[0K    -- Setting call duration limit to 5088000.000 seconds.

[Kserver066*CLI>
[0K  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6

[Kserver066*CLI>
[0KAudio is at 77.77.77.66 port 13810
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 77.77.77.88:5060:
INVITE sip:441234567890@77.77.77.88 SIP/2.0Via: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK4b0bed95;rportMax-Forwards: 70From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>Contact: <sip:441234567890@77.77.77.66>Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 INVITEUser-Agent: Asterisk PBX 1.6.2.24Date: Sun, 11 Nov 2012 16:24:41 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Type: application/sdpContent-Length: 261v=0o=root 1044909343 1044909343 IN IP4 77.77.77.66s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.66t=0 0m=audio 13810 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
---

[Kserver066*CLI>
[0K    -- Called A2B-SECOND-SERVER/441234567890

[Kserver066*CLI>
[0K
<--- SIP read from UDP:77.77.77.88:5060 --->
SIP/2.0 100 TryingVia: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK4b0bed95;received=77.77.77.66;rport=5060From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.88>Content-Length: 0
<------------->
--- (11 headers 0 lines) ---

[Kserver066*CLI>
[0K
<--- SIP read from UDP:77.77.77.88:5060 --->
SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK4b0bed95;received=77.77.77.66;rport=5060From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>;tag=as0e885c15Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.88>Content-Type: application/sdpContent-Length: 261v=0o=root 2063983537 2063983537 IN IP4 77.77.77.88s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.88t=0 0m=audio 10016 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 8
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 telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 77.77.77.88:10016
    -- SIP/A2B-SECOND-SERVER-00000005 is making progress passing it to SIP/0024476691-00000004
Audio is at 77.77.77.66 port 13634
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 1.1.1.1:53450 --->
SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1836708136;received=1.1.1.1From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>;tag=as2decf668Call-ID: 1953639537@192.168.10.1CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.66>Content-Type: application/sdpContent-Length: 302v=0o=root 9081837 9081837 IN IP4 77.77.77.66s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.66t=0 0m=audio 13634 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
<------------>

[Kserver066*CLI>
[0K
<--- SIP read from UDP:77.77.77.88:5060 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK4b0bed95;received=77.77.77.66;rport=5060From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>;tag=as0e885c15Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.88>Content-Type: application/sdpContent-Length: 261v=0o=root 2063983537 2063983538 IN IP4 77.77.77.88s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.88t=0 0m=audio 10016 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 8
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 telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 77.77.77.88:10016
list_route: hop: <sip:441234567890@77.77.77.88>
set_destination: Parsing <sip:441234567890@77.77.77.88> for address/port to send to
set_destination: set destination to 77.77.77.88, port 5060
Transmitting (no NAT) to 77.77.77.88:5060:
ACK sip:441234567890@77.77.77.88 SIP/2.0Via: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK3baec434;rportMax-Forwards: 70From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>;tag=as0e885c15Contact: <sip:441234567890@77.77.77.66>Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 ACKUser-Agent: Asterisk PBX 1.6.2.24Content-Length: 0
---
    -- SIP/A2B-SECOND-SERVER-00000005 answered SIP/0024476691-00000004
Audio is at 77.77.77.66 port 13634
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 1.1.1.1:53450 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1836708136;received=1.1.1.1From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859
[Kserver066*CLI>
[0KTo: <sip:441234567890@77.77.77.66>;tag=as2decf668Call-ID: 1953639537@192.168.10.1CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.66>Content-Type: application/sdpContent-Length: 302v=0o=root 9081837 9081838 IN IP4 77.77.77.66s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.66t=0 0m=audio 13634 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
<------------>
    -- Packet2Packet bridging SIP/0024476691-00000004 and SIP/A2B-SECOND-SERVER-00000005

[Kserver066*CLI>
[0K
<--- SIP read from UDP:77.77.77.88:5060 --->
BYE sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 77.77.77.88:5060;branch=z9hG4bK38db1bd6;rportMax-Forwards: 70From: <sip:441234567890@77.77.77.88>;tag=as0e885c15To: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eCall-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 BYEUser-Agent: Asterisk PBX 1.6.2.24X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 77.77.77.88 : 5060 (no NAT)

<--- Transmitting (no NAT) to 77.77.77.88:5060 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 77.77.77.88:5060;branch=z9hG4bK38db1bd6;received=77.77.77.88;rport=5060From: <sip:441234567890@77.77.77.88>;tag=as0e885c15To: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eCall-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 BYEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0
<------------>

[Kserver066*CLI>
[0K    -- <SIP/0024476691-00000004>AGI Script a2billing.php completed, returning 4

[Kserver066*CLI>
[0K  == Spawn extension (a2billing, 441234567890, 1) exited non-zero on 'SIP/0024476691-00000004'

[Kserver066*CLI>
[0KScheduling destruction of SIP dialog '1953639537@192.168.10.1' in 32000 ms (Method: INVITE)

[Kserver066*CLI>
[0K
<--- SIP read from UDP:1.1.1.1:53450 --->
ACK sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK553523559From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>;tag=as2decf668Call-ID: 1953639537@192.168.10.1CSeq: 2 ACKContact: <sip:0024476691@192.168.10.1:5063>Max-Forwards: 70User-Agent: Yealink SIP-T26P 6.61.0.83Content-Length: 0
<------------->
--- (10 headers 0 lines) ---

[Kserver066*CLI>
[0Kset_destination: Parsing <sip:0024476691@192.168.10.1:5063> for address/port to send to
set_destination: set destination to 192.168.10.1, port 5063
Reliably Transmitting (NAT) to 1.1.1.1:53450:
BYE sip:0024476691@192.168.10.1:5063 SIP/2.0Via: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK427777eb;rportMax-Forwards: 70From: <sip:441234567890@77.77.77.66>;tag=as2decf668To: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859Call-ID: 1953639537@192.168.10.1CSeq: 102 BYEUser-Agent: Asterisk PBX 1.6.2.24X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
---
Scheduling destruction of SIP dialog '1953639537@192.168.10.1' in 32000 ms (Method: ACK)
Really destroying SIP dialog '47bb4c505b70e695224506336ef57238@77.77.77.66' Method: BYE

[Kserver066*CLI>
[0K
<--- SIP read from UDP:1.1.1.1:53450 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK427777eb;rportFrom: <sip:441234567890@77.77.77.66>;tag=as2decf668To: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859Call-ID: 1953639537@192.168.10.1CSeq: 102 BYEUser-Agent: Yealink SIP-T26P 6.61.0.83Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1953639537@192.168.10.1' Method: ACK

[Kserver066*CLI> exitExecuting last minute cleanups

============(CALL DROPS HERE)=============


Asterisk ending (0).
[0m]0;root@server066:/etc/asterisk[root@server066 asterisk]#