
<------------>
    -- Executing [1000@outbound:1] Macro("SIP/2000-00000047", "callrin,1000") in new stack
    -- Executing [s@macro-callrin:1] Set("SIP/2000-00000047", "CALLERID(name)=1000--2000") in new stack
    -- Executing [s@macro-callrin:2] Set("SIP/2000-00000047", "CDR(accountcode)=INBOUND") in new stack
    -- Executing [s@macro-callrin:3] Set("SIP/2000-00000047", "date=2017-08-07") in new stack
    -- Executing [s@macro-callrin:4] Set("SIP/2000-00000047", "time=12-06-31") in new stack
    -- Executing [s@macro-callrin:5] Set("SIP/2000-00000047", "MONITOR_FILENAME=/var/spool/asterisk/monitor/2017-08-07/INBOUND/1000--2000---12-06-31") in new stack
    -- Executing [s@macro-callrin:6] Monitor("SIP/2000-00000047", "wav,/var/spool/asterisk/monitor/2017-08-07/INBOUND/1000--2000---12-06-31,m") in new stack
    -- Executing [1000@outbound:2] Dial("SIP/2000-00000047", "SIP/1000,20,Tt") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14766
Adding codec 1000008 (g729) to SDP
Adding codec 1000001 (g723) to SDP
Adding codec 1000003 (ulaw) to SDP
Adding codec 1000002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 182.176.xxx.xxx:1188:
INVITE sip:1000@182.176.xxx.xxx:1191;transport=udp SIP/2.0
Via: SIP/2.0/UDP 194.88.xxx.xxx:5060;branch=z9hG4bK012fc7e7;rport
Max-Forwards: 70
From: "1000--2000" <sip:2000@194.88.xxx.xxx>;tag=as51ab6806
To: <sip:1000@182.176.xxx.xxx:1191;transport=udp>
Contact: <sip:2000@194.88.xxx.xxx:5060>
Call-ID: 79eed95a31434bd6233283ae2a9cab5b@194.88.xxx.xxx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.1
Date: Mon, 07 Aug 2017 16:06:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 348

v=0
o=root 175501252 175501252 IN IP4 194.88.xxx.xxx
s=Asterisk PBX 11.25.1
c=IN IP4 194.88.xxx.xxx
t=0 0
m=audio 14766 RTP/AVP 18 4 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/1000

<--- SIP read from UDP:182.176.xxx.xxx:1188 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 194.88.xxx.xxx:5060;branch=z9hG4bK012fc7e7;rport
From: "1000--2000" <sip:2000@194.88.xxx.xxx>;tag=as51ab6806
To: <sip:1000@182.176.xxx.xxx:1191;transport=udp>
Call-ID: 79eed95a31434bd6233283ae2a9cab5b@194.88.xxx.xxx:5060
Date: Mon, 07 Aug 2017 14:06:55 GMT
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 182.176.xxx.xxx:1188:
ACK sip:1000@182.176.xxx.xxx:1191;transport=udp SIP/2.0
Via: SIP/2.0/UDP 194.88.xxx.xxx:5060;branch=z9hG4bK012fc7e7;rport
Max-Forwards: 70
From: "1000--2000" <sip:2000@194.88.xxx.xxx>;tag=as51ab6806
To: <sip:1000@182.176.xxx.xxx:1191;transport=udp>
Contact: <sip:2000@194.88.xxx.xxx:5060>
Call-ID: 79eed95a31434bd6233283ae2a9cab5b@194.88.xxx.xxx:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '79eed95a31434bd6233283ae2a9cab5b@194.88.xxx.xxx:5060' in 16384 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1000@outbound:3] Macro("SIP/2000-00000047", "vmtech,1000") in new stack
    -- Executing [s@macro-vmtech:1] VoiceMail("SIP/2000-00000047", "1000@vmtech") in new stack
Audio is at 19276
Adding codec 1000008 (g729) to SDP
Adding codec 1000003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 182.176.xxx.xxx:29373 --->
SIP/2.0 2000 OK
Via: SIP/2.0/UDP 192.168.10.40:1496;branch=z9hG4bK78c80266;received=182.176.xxx.xxx;rport=29373
From: "2000" <sip:2000@194.88.xxx.xxx>;tag=0015f9dd75dab7591b0b97ed-28d8698b
To: <sip:1000@194.88.xxx.xxx>;tag=as79f50cfd
Call-ID: 0015f9dd-75da008d-36217f27-0059de1b@192.168.11.106
CSeq: 102 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1000@194.88.xxx.xxx:5060>
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2114304497 2114304497 IN IP4 194.88.xxx.xxx
s=Asterisk PBX 11.25.1
c=IN IP4 194.88.xxx.xxx
t=0 0
m=audio 19276 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv