Untitled

sip.cfg from the applied template - Polycome IP335 TEST:

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- PlcmConversionCreatedFile version=1.2 converted=Tue Oct 25 13:12:50 2011 -->
<polycomConfig xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="polycomConfig.xsd">
  <se>
    <se.rt>
      <se.rt.custom1 se.rt.custom1.name="Low Double Trill" se.rt.custom1.ringer="ringer3" />
      <se.rt.custom2 se.rt.custom2.name="Medium Trill" se.rt.custom2.ringer="ringer4" />
      <se.rt.custom3 se.rt.custom3.name="Triplet" se.rt.custom3.ringer="ringer11" />
    </se.rt>
  </se>
  <voIpProt>
    <voIpProt.SIP>
      <voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.value="internal" voIpProt.SIP.alertInfo.1.class="custom1"/>
      <voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.2.value="external" voIpProt.SIP.alertInfo.2.class="custom2"/>
      <voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.3.value="directory" voIpProt.SIP.alertInfo.3.class="custom3"/>
    </voIpProt.SIP>
  </voIpProt>
</polycomConfig>


Inbound route to 01142189188 from 07871345768 using Alert Info <external>:


login as: root
root@192.168.210.12's password:
Last login: Tue Aug 25 16:38:32 2015 from 192.168.210.117


 _____              ____  ______  __
|  ___| __ ___  ___|  _ \| __ ) \/ /
| |_ | '__/ _ \/ _ \ |_) |  _ \  /
|  _|| | |  __/  __/  __/| |_) /  \
|_|  |_|  \___|\___|_|   |____/_/\_\

Interface eth0 IP: 192.168.210.12

Please note most tasks should be handled through the FreePBX UI.
You can access the FreePBX GUI by typing one of the above IP's in to your web browser.
For support please visit http://www.freepbx.org/support-and-professional-services

[root@localhost ~]# asterisk -rvvvvvvvv
Asterisk 11.4.0, Copyright (C) 1999 - 2012 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 11.4.0 currently running on localhost (pid = 2066)
localhost*CLI> sip set debug on
SIP Debugging re-enabled

<--- SIP read from UDP:83.166.160.240:5060 --->
INVITE sip:01142189188@213.123.58.246 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142189188@213.123.58.246>
From: <sip:07871345768@83.166.160.240>;tag=3649506011-334677
Call-ID: 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 83.166.160.240:5060;branch=z9hG4bKb9766445a29614572a87ba45cc775da8
Contact: <sip:07871345768@83.166.160.240:5060>
Expires: 300
Content-Type: application/sdp
Accept: application/sdp
User-Agent: Node4
Content-Length: 350

v=0
o=N4-DRY-NXT-SBS-01 59627398 0 IN IP4 83.166.160.240
s=sip call
c=IN IP4 83.166.160.241
t=0 0
m=audio 41050 RTP/AVP 0 18 8 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
<------------->
--- (17 headers 15 lines) ---
Sending to 83.166.160.240:5060 (NAT)
Sending to 83.166.160.240:5060 (NAT)
Using INVITE request as basis request - 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
Found peer 'Node4' for '07871345768' from 83.166.160.240:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 4
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 G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 83.166.160.241:41050
Looking for 01142189188 in from-trunk-sip-Node4 (domain 213.123.58.246)
list_route: hop: <sip:07871345768@83.166.160.240:5060>

<--- Transmitting (NAT) to 83.166.160.240:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.166.160.240:5060;branch=z9hG4bKb9766445a29614572a87ba45cc775da8;received=83.166.160.240;rport=5060
From: <sip:07871345768@83.166.160.240>;tag=3649506011-334677
To: <sip:01142189188@213.123.58.246>
Call-ID: 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:01142189188@213.123.58.246:5060>
Content-Length: 0


<------------>
    -- Executing [01142189188@from-trunk-sip-Node4:1] Set("SIP/Node4-0000d914", "GROUP()=OUT_1") in new stack
    -- Executing [01142189188@from-trunk-sip-Node4:2] Goto("SIP/Node4-0000d914", "from-trunk,01142189188,1") in new stack
    -- Goto (from-trunk,01142189188,1)
    -- Executing [01142189188@from-trunk:1] Set("SIP/Node4-0000d914", "__FROM_DID=01142189188") in new stack
    -- Executing [01142189188@from-trunk:2] Gosub("SIP/Node4-0000d914", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/Node4-0000d914", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/Node4-0000d914", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/Node4-0000d914", "") in new stack
    -- Executing [01142189188@from-trunk:3] Set("SIP/Node4-0000d914", "CDR(did)=01142189188") in new stack
    -- Executing [01142189188@from-trunk:4] ExecIf("SIP/Node4-0000d914", "1 ?Set(CALLERID(name)=07871345768)") in new stack
    -- Executing [01142189188@from-trunk:5] Set("SIP/Node4-0000d914", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [01142189188@from-trunk:6] Set("SIP/Node4-0000d914", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [01142189188@from-trunk:7] Set("SIP/Node4-0000d914", "__ALERT_INFO=<external>") in new stack
    -- Executing [01142189188@from-trunk:8] Goto("SIP/Node4-0000d914", "from-did-direct,6410,1") in new stack
    -- Goto (from-did-direct,6410,1)
    -- Executing [6410@from-did-direct:1] Set("SIP/Node4-0000d914", "__RINGTIMER=15") in new stack
    -- Executing [6410@from-did-direct:2] Macro("SIP/Node4-0000d914", "exten-vm,6410,6410,0,0,0") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/Node4-0000d914", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/Node4-0000d914", "TOUCH_MONITOR=1440517211.170303") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/Node4-0000d914", "AMPUSER=07871345768") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/Node4-0000d914", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/Node4-0000d914", "1?Set(REALCALLERIDNUM=07871345768)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/Node4-0000d914", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:6] Set("SIP/Node4-0000d914", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:7] GotoIf("SIP/Node4-0000d914", "1?report") in new stack
    -- Goto (macro-user-callerid,s,14)
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/Node4-0000d914", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:15] Set("SIP/Node4-0000d914", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:16] GotoIf("SIP/Node4-0000d914", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,27)
    -- Executing [s@macro-user-callerid:27] Set("SIP/Node4-0000d914", "CALLERID(number)=07871345768") in new stack
    -- Executing [s@macro-user-callerid:28] Set("SIP/Node4-0000d914", "CALLERID(name)=07871345768") in new stack
    -- Executing [s@macro-user-callerid:29] Set("SIP/Node4-0000d914", "CDR(cnum)=07871345768") in new stack
    -- Executing [s@macro-user-callerid:30] Set("SIP/Node4-0000d914", "CDR(cnam)=07871345768") in new stack
    -- Executing [s@macro-user-callerid:31] Set("SIP/Node4-0000d914", "CHANNEL(language)=uk-female") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/Node4-0000d914", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/Node4-0000d914", "__EXTTOCALL=6410") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/Node4-0000d914", "__PICKUPMARK=6410") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/Node4-0000d914", "RT=15") in new stack
    -- Executing [s@macro-exten-vm:6] ExecIf("SIP/Node4-0000d914", "0?Macro(vm,6410,DIRECTDIAL,)") in new stack
    -- Executing [s@macro-exten-vm:7] ExecIf("SIP/Node4-0000d914", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:8] Gosub("SIP/Node4-0000d914", "sub-record-check,s,1(exten,6410,)") in new stack
    -- Executing [s@sub-record-check:1] Set("SIP/Node4-0000d914", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:2] GotoIf("SIP/Node4-0000d914", "1?check") in new stack
    -- Goto (sub-record-check,s,7)
    -- Executing [s@sub-record-check:7] Set("SIP/Node4-0000d914", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:8] GotoIf("SIP/Node4-0000d914", "1?next") in new stack
    -- Goto (sub-record-check,s,11)
    -- Executing [s@sub-record-check:11] ExecIf("SIP/Node4-0000d914", "0?Return()") in new stack
    -- Executing [s@sub-record-check:12] ExecIf("SIP/Node4-0000d914", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [s@sub-record-check:13] GotoIf("SIP/Node4-0000d914", "0?exten,1") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/Node4-0000d914", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/Node4-0000d914", "NOW=1440517211") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/Node4-0000d914", "__DAY=25") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/Node4-0000d914", "__MONTH=08") in new stack
    -- Executing [s@sub-record-check:18] Set("SIP/Node4-0000d914", "__YEAR=2015") in new stack
    -- Executing [s@sub-record-check:19] Set("SIP/Node4-0000d914", "__TIMESTR=20150825-164011") in new stack
    -- Executing [s@sub-record-check:20] Set("SIP/Node4-0000d914", "__FROMEXTEN=07871345768") in new stack
    -- Executing [s@sub-record-check:21] Set("SIP/Node4-0000d914", "__CALLFILENAME=exten-6410-07871345768-20150825-164011-1440517211.170303") in new stack
    -- Executing [s@sub-record-check:22] Goto("SIP/Node4-0000d914", "exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten@sub-record-check:1] GotoIf("SIP/Node4-0000d914", "0?callee") in new stack
    -- Executing [exten@sub-record-check:2] Set("SIP/Node4-0000d914", "__REC_POLICY_MODE=always") in new stack
    -- Executing [exten@sub-record-check:3] GotoIf("SIP/Node4-0000d914", "0?caller") in new stack
    -- Executing [exten@sub-record-check:4] GotoIf("SIP/Node4-0000d914", "1?callee") in new stack
    -- Goto (sub-record-check,exten,8)
    -- Executing [exten@sub-record-check:8] GosubIf("SIP/Node4-0000d914", "1?record,1(exten,6410,07871345768)") in new stack
    -- Executing [record@sub-record-check:1] Set("SIP/Node4-0000d914", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
    -- Executing [record@sub-record-check:2] MixMonitor("SIP/Node4-0000d914", "2015/08/25/exten-6410-07871345768-20150825-164011-1440517211.170303.wav,,") in new stack
    -- Executing [record@sub-record-check:3] Set("SIP/Node4-0000d914", "__REC_STATUS=RECORDING") in new stack
    -- Executing [record@sub-record-check:4] Set("SIP/Node4-0000d914", "CDR(recordingfile)=exten-6410-07871345768-20150825-164011-1440517211.170303.wav") in new stack
    -- Executing [record@sub-record-check:5] Return("SIP/Node4-0000d914", "") in new stack
    -- Executing [exten@sub-record-check:9] Return("SIP/Node4-0000d914", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/Node4-0000d914", "dial-one,15,Ttr,6410") in new stack
    -- Executing [s@macro-dial-one:1] Set("SIP/Node4-0000d914", "DEXTEN=6410") in new stack
    -- Executing [s@macro-dial-one:2] Set("SIP/Node4-0000d914", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:3] GosubIf("SIP/Node4-0000d914", "0?screen,1()") in new stack
    -- Executing [s@macro-dial-one:4] GosubIf("SIP/Node4-0000d914", "0?cf,1()") in new stack
    -- Executing [s@macro-dial-one:5] GotoIf("SIP/Node4-0000d914", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,8)
    -- Executing [s@macro-dial-one:8] GotoIf("SIP/Node4-0000d914", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:9] GotoIf("SIP/Node4-0000d914", "0?continue") in new stack
    -- Executing [s@macro-dial-one:10] Set("SIP/Node4-0000d914", "EXTHASCW=") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("SIP/Node4-0000d914", "1?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,12)
    -- Executing [s@macro-dial-one:12] GotoIf("SIP/Node4-0000d914", "0?docfu:skip3") in new stack
    -- Goto (macro-dial-one,s,16)
    -- Executing [s@macro-dial-one:16] GotoIf("SIP/Node4-0000d914", "1?next2:continue") in new stack
    -- Goto (macro-dial-one,s,17)
    -- Executing [s@macro-dial-one:17] GotoIf("SIP/Node4-0000d914", "1?continue") in new stack
    -- Goto (macro-dial-one,s,25)
    -- Executing [s@macro-dial-one:25] GotoIf("SIP/Node4-0000d914", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:26] GosubIf("SIP/Node4-0000d914", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("SIP/Node4-0000d914", "DSTRING=") in new stack
  == Begin MixMonitor Recording SIP/Node4-0000d914
    -- Executing [dstring@macro-dial-one:2] Set("SIP/Node4-0000d914", "DEVICES=6410") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/Node4-0000d914", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/Node4-0000d914", "0?Set(DEVICES=410)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("SIP/Node4-0000d914", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("SIP/Node4-0000d914", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("SIP/Node4-0000d914", "THISDIAL=SIP/6410") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/Node4-0000d914", "1?zap2dahdi,1()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/Node4-0000d914", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/Node4-0000d914", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/Node4-0000d914", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/Node4-0000d914", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/Node4-0000d914", "THISPART2=SIP/6410") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/Node4-0000d914", "0?Set(THISPART2=DAHDI/6410)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/Node4-0000d914", "NEWDIAL=SIP/6410&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/Node4-0000d914", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/Node4-0000d914", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/Node4-0000d914", "THISDIAL=SIP/6410") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/Node4-0000d914", "") in new stack
    -- Executing [dstring@macro-dial-one:9] Set("SIP/Node4-0000d914", "DSTRING=SIP/6410&") in new stack
    -- Executing [dstring@macro-dial-one:10] Set("SIP/Node4-0000d914", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/Node4-0000d914", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:12] Set("SIP/Node4-0000d914", "DSTRING=SIP/6410") in new stack
    -- Executing [dstring@macro-dial-one:13] Return("SIP/Node4-0000d914", "") in new stack
    -- Executing [s@macro-dial-one:27] GotoIf("SIP/Node4-0000d914", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GotoIf("SIP/Node4-0000d914", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:29] GosubIf("SIP/Node4-0000d914", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("SIP/Node4-0000d914", "DB(CALLTRACE/6410)=07871345768") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("SIP/Node4-0000d914", "") in new stack
    -- Executing [s@macro-dial-one:30] Set("SIP/Node4-0000d914", "D_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dial-one:31] ExecIf("SIP/Node4-0000d914", "1?SIPAddHeader(Alert-Info: <external>)") in new stack
    -- Executing [s@macro-dial-one:32] ExecIf("SIP/Node4-0000d914", "0?SIPAddHeader()") in new stack
    -- Executing [s@macro-dial-one:33] ExecIf("SIP/Node4-0000d914", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s@macro-dial-one:34] GosubIf("SIP/Node4-0000d914", "0?qwait,1()") in new stack
    -- Executing [s@macro-dial-one:35] Set("SIP/Node4-0000d914", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:36] Set("SIP/Node4-0000d914", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:37] GotoIf("SIP/Node4-0000d914", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:38] GotoIf("SIP/Node4-0000d914", "1?godial") in new stack
    -- Goto (macro-dial-one,s,42)
    -- Executing [s@macro-dial-one:42] Dial("SIP/Node4-0000d914", "SIP/6410,15,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 11904
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.210.141:5060:
INVITE sip:6410@192.168.210.141:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK7ca8adf3
Max-Forwards: 70
From: "07871345768" <sip:07871345768@192.168.210.12>;tag=as45e109cb
To: <sip:6410@192.168.210.141:5060>
Contact: <sip:07871345768@192.168.210.12:5060>
Call-ID: 28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Date: Tue, 25 Aug 2015 15:40:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Alert-Info: <external>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 346030699 346030699 IN IP4 192.168.210.12
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.210.12
t=0 0
m=audio 11904 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/6410

<--- Transmitting (NAT) to 83.166.160.240:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.166.160.240:5060;branch=z9hG4bKb9766445a29614572a87ba45cc775da8;received=83.166.160.240;rport=5060
From: <sip:07871345768@83.166.160.240>;tag=3649506011-334677
To: <sip:01142189188@213.123.58.246>;tag=as454eefe6
Call-ID: 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:01142189188@213.123.58.246:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.210.141:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK7ca8adf3
From: "07871345768" <sip:07871345768@192.168.210.12>;tag=as45e109cb
To: "6410" <sip:6410@192.168.210.141:5060>;tag=716AC414-9367E2C1
CSeq: 102 INVITE
Call-ID: 28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060
Contact: <sip:6410@192.168.210.141:5060>
User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.0.2.11307
Accept-Language: en
Content-Length: 0

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

<--- SIP read from UDP:192.168.210.141:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK7ca8adf3
From: "07871345768" <sip:07871345768@192.168.210.12>;tag=as45e109cb
To: "6410" <sip:6410@192.168.210.141:5060>;tag=716AC414-9367E2C1
CSeq: 102 INVITE
Call-ID: 28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060
Contact: <sip:6410@192.168.210.141:5060>
User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.0.2.11307
Allow-Events: conference,talk,hold
Accept-Language: en
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:6410@192.168.210.141:5060>
    -- SIP/6410-0000d915 is ringing

<--- Transmitting (NAT) to 83.166.160.240:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.166.160.240:5060;branch=z9hG4bKb9766445a29614572a87ba45cc775da8;received=83.166.160.240;rport=5060
From: <sip:07871345768@83.166.160.240>;tag=3649506011-334677
To: <sip:01142189188@213.123.58.246>;tag=as454eefe6
Call-ID: 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:01142189188@213.123.58.246:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.210.141:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK7ca8adf3
From: "07871345768" <sip:07871345768@192.168.210.12>;tag=as45e109cb
To: "6410" <sip:6410@192.168.210.141:5060>;tag=716AC414-9367E2C1
CSeq: 102 INVITE
Call-ID: 28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060
Contact: <sip:6410@192.168.210.141:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.0.2.11307
Accept-Language: en
Content-Type: application/sdp
Content-Length: 217

v=0
o=- 1440517119 1440517119 IN IP4 192.168.210.141
s=Polycom IP Phone
c=IN IP4 192.168.210.141
t=0 0
a=sendrecv
m=audio 2236 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (13 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.210.141:2236
list_route: hop: <sip:6410@192.168.210.141:5060>
set_destination: Parsing <sip:6410@192.168.210.141:5060> for address/port to send to
set_destination: set destination to 192.168.210.141:5060
Transmitting (no NAT) to 192.168.210.141:5060:
ACK sip:6410@192.168.210.141:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK0fbc4549
Max-Forwards: 70
From: "07871345768" <sip:07871345768@192.168.210.12>;tag=as45e109cb
To: <sip:6410@192.168.210.141:5060>;tag=716AC414-9367E2C1
Contact: <sip:07871345768@192.168.210.12:5060>
Call-ID: 28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.4.0)
Content-Length: 0


---
    -- SIP/6410-0000d915 answered SIP/Node4-0000d914
Audio is at 13774
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 83.166.160.240:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.166.160.240:5060;branch=z9hG4bKb9766445a29614572a87ba45cc775da8;received=83.166.160.240;rport=5060
From: <sip:07871345768@83.166.160.240>;tag=3649506011-334677
To: <sip:01142189188@213.123.58.246>;tag=as454eefe6
Call-ID: 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:01142189188@213.123.58.246:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 261

v=0
o=root 275953746 275953746 IN IP4 213.123.58.246
s=Asterisk PBX 11.4.0
c=IN IP4 213.123.58.246
t=0 0
m=audio 13774 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:83.166.160.240:5060 --->
ACK sip:01142189188@213.123.58.246:5060 SIP/2.0
Max-Forwards: 69
To: <sip:01142189188@213.123.58.246>;tag=as454eefe6
From: <sip:07871345768@83.166.160.240>;tag=3649506011-334677
Call-ID: 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
CSeq: 1 ACK
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 83.166.160.240:5060;branch=z9hG4bK0bf5e539e4dabc50fcedf3fb20bcfd70
Contact: <sip:07871345768@83.166.160.240:5060>
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
       > 0xb3bbbe98 -- Probation passed - setting RTP source address to 192.168.210.141:2236
       > 0xb7729558 -- Probation passed - setting RTP source address to 83.166.160.241:41050
Reliably Transmitting (no NAT) to 192.168.210.146:5060:
OPTIONS sip:6417@192.168.210.146:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK332255aa
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.210.12>;tag=as71945b2e
To: <sip:6417@192.168.210.146:5060>
Contact: <sip:Unknown@192.168.210.12:5060>
Call-ID: 20e9b3472e0d1d4b508ec80117d77bae@192.168.210.12:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.4.0)
Date: Tue, 25 Aug 2015 15:40:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.210.146:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK332255aa
From: "Unknown" <sip:Unknown@192.168.210.12>;tag=as71945b2e
To: <sip:6417@192.168.210.146:5060>;tag=ar60854c3d
Call-ID: 20e9b3472e0d1d4b508ec80117d77bae@192.168.210.12:5060
CSeq: 102 OPTIONS
Supported: replaces
User-Agent: N300 IP/42.076.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '20e9b3472e0d1d4b508ec80117d77bae@192.168.210.12:5060' Method: OPTIONS

<--- SIP read from UDP:83.166.160.240:5060 --->
BYE sip:01142189188@213.123.58.246:5060 SIP/2.0
Max-Forwards: 69
To: <sip:01142189188@213.123.58.246>;tag=as454eefe6
From: <sip:07871345768@83.166.160.240>;tag=3649506011-334677
Call-ID: 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
CSeq: 2 BYE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 83.166.160.240:5060;branch=z9hG4bKb7c1a29571715dc731f84bc40783d77d
Contact: <sip:07871345768@83.166.160.240:5060>
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 83.166.160.240:5060 (NAT)
Scheduling destruction of SIP dialog '37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 83.166.160.240:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.166.160.240:5060;branch=z9hG4bKb7c1a29571715dc731f84bc40783d77d;received=83.166.160.240;rport=5060
From: <sip:07871345768@83.166.160.240>;tag=3649506011-334677
To: <sip:01142189188@213.123.58.246>;tag=as454eefe6
Call-ID: 37400675-3649506011-334673@N4-DRY-NXT-SBS-01.node4.net
CSeq: 2 BYE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [h@macro-dial-one:1] Macro("SIP/Node4-0000d914", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/Node4-0000d914", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/Node4-0000d914", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("SIP/Node4-0000d914", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/Node4-0000d914' in macro 'hangupcall'
  == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/Node4-0000d914'
Scheduling destruction of SIP dialog '28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:6410@192.168.210.141:5060> for address/port to send to
set_destination: set destination to 192.168.210.141:5060
Reliably Transmitting (no NAT) to 192.168.210.141:5060:
BYE sip:6410@192.168.210.141:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK277c11a2
Max-Forwards: 70
From: "07871345768" <sip:07871345768@192.168.210.12>;tag=as45e109cb
To: <sip:6410@192.168.210.141:5060>;tag=716AC414-9367E2C1
Call-ID: 28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.4.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/Node4-0000d914' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/Node4-0000d914' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 6410, 2) exited non-zero on 'SIP/Node4-0000d914'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/Node4-0000d914

<--- SIP read from UDP:192.168.210.141:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.210.12:5060;branch=z9hG4bK277c11a2
From: "07871345768" <sip:07871345768@192.168.210.12>;tag=as45e109cb
To: "6410" <sip:6410@192.168.210.141:5060>;tag=716AC414-9367E2C1
CSeq: 103 BYE
Call-ID: 28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060
Contact: <sip:6410@192.168.210.141:5060>
User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.0.2.11307
Accept-Language: en
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '28f46ca11e3a9878658f3c1a4ece4c68@192.168.210.12:5060' Method: INVITE
localhost*CLI>