AMD 8369 issue

[ed123]

Hi,
We test an active number and it work in autodial with 8368 extension. But when i activate the answering machine detection(8369) the call drop after i pick it up. THe AMD tag it as NA.

Logs:

[Aug 30 07:06:20] == Using SIP RTP CoS mark 5
[Aug 30 07:06:20] -- Called SIP/MYPHONENUMBER@VOIP
[Aug 30 07:06:22] > 0x7fc27c0afb50 -- Strict RTP learning after remote address set to: 192.170.159.196:12640
[Aug 30 07:06:22] -- SIP/VOIP-00000004 is making progress passing it to Local/9MYPHONENUMBER@default-00000004;2
[Aug 30 07:06:25] -- SIP/VOIP-00000004 answered Local/9MYPHONENUMBER@default-00000004;2
[Aug 30 07:06:25] > Channel Local/9MYPHONENUMBER@default-00000004;1 was answered
[Aug 30 07:06:25] -- Executing [8369@default:1] Playback("Local/9MYPHONENUMBER@default-00000004;1", "sip-silence") in new stack
[Aug 30 07:06:25] -- <Local/9MYPHONENUMBER@default-00000004;1> Playing 'sip-silence.gsm' (language 'en')
[Aug 30 07:06:25] -- Executing [8369@default:2] AGI("Local/9MYPHONENUMBER@default-00000004;1", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 30 07:06:25] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=T3PH))
[Aug 30 07:06:25] -- <Local/9MYPHONENUMBER@default-00000004;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 30 07:06:25] -- Executing [8369@default:3] AMD("Local/9MYPHONENUMBER@default-00000004;1", "2000,2000,1000,5000,120,50,4,256") in new stack
[Aug 30 07:06:25] -- AMD: Local/9MYPHONENUMBER@default-00000004;1 14046927964 (N/A) (Fmt: slin)
[Aug 30 07:06:25] -- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256] maximumWordLength [5000]
[Aug 30 07:06:25] -- AMD: Channel [SIP/VOIP-00000004]. Changed state to STATE_IN_SILENCE
[Aug 30 07:06:25] -- Executing [h@default:1] AGI("Local/9MYPHONENUMBER@default-00000004;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----0") in new stack
[Aug 30 07:06:26] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 30 07:06:26] -- <Local/9MYPHONENUMBER@default-00000004;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---5-----0 completed, returning 0
[Aug 30 07:06:26] == Spawn extension (default, 9MYPHONENUMBER, 2) exited non-zero on 'Local/9MYPHONENUMBER@default-00000004;2'
[Aug 30 07:06:27] -- AMD: Channel [SIP/VOIP-00000004]. ANSWERING MACHINE: silenceDuration:2000 initialSilence:2000
[Aug 30 07:06:27] -- Executing [8369@default:4] AGI("SIP/VOIP-00000004", "VD_amd.agi,8369") in new stack
[Aug 30 07:06:27] -- Launched AGI Script /usr/share/asterisk/agi-bin/VD_amd.agi
[Aug 30 07:06:27] -- <SIP/VOIP-00000004> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 30 07:06:27] -- <SIP/VOIP-00000004> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 30 07:06:27] -- <SIP/VOIP-00000004>AGI Script VD_amd.agi completed, returning 4
[Aug 30 07:06:27] == Spawn extension (default, 8369, 4) exited non-zero on 'SIP/VOIP-00000004'
[Aug 30 07:06:27] -- Executing [h@default:1] AGI("SIP/VOIP-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 30 07:06:27] -- <SIP/VOIP-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 30 07:06:38] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 30 07:06:38] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/1001-00000003
[Aug 30 07:06:38] -- Hungup 'DAHDI/pseudo-974560985'
[Aug 30 07:06:38] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/1001-00000003'
[Aug 30 07:06:38] -- Executing [h@default:1] AGI("SIP/1001-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Aug 30 07:06:38] -- <SIP/1001-00000003>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 30 07:06:38] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 30 07:06:38] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000005;2", "8600051,K") in new stack
[Aug 30 07:06:38] WARNING[8821][C-00000007]: app_meetme.c:5053 admin_exec: Conference number '8600051' not found!
[Aug 30 07:06:38] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000005;2", "") in new stack
[Aug 30 07:06:38] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000005;2'
[Aug 30 07:06:38] -- Executing [h@default:1] AGI("Local/55558600051@default-00000005;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Aug 30 07:06:38] -- <Local/55558600051@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 30 07:06:39] == Manager 'sendcron' logged off from 127.0.0.1

[ed123]

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,Playback(sip-silence)
exten => 8369,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,n,AMD(2000,2000,1000,5000,120,50,4,256)
exten => 8369,n,AGI(VD_amd.agi,${EXTEN})
exten => 8369,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,n,Hangup()


Account Entry:
[VOIP]
type=friend
user=user
username=user
fromuser=user
authname=user
secret=password
host=voip.net
nat=force_rport,comedia
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=g726
allow=g729
canreinvite=no
insecure=very
dtmfmode=rfc2833
context=trunkinbound

Dialplan:

exten => _91X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91X.,n,Dial(SIP/${EXTEN:1}@VOIP,60,tToR)
exten => _91X.,n,Hangup

VERSION: 2.14-685a
BUILD: 180825-2100
© 2018 ViciDial Group
ViciBox_v8.x86_64-8.0.1.iso
No other hardware installed
G729
Local Network only

[blackbird2306]

What is the real problem with your issue? The DROP after pickup (which is normal in your case) or the fact that the status is not "AA" but "NA"?

[ed123]

(8369)
the issue is when i pickup the call from myphone no voice and call hungup after 1-2 secs., on agent side the call did not pass(after i pickup) and nothing happens. The agent just sit there waiting for a call.

(8368)
when i pickup the call from myphone, the call pass directly to the agent with no issue.

[ed123]

the call pickup is a live call supposedly but when i activate the AMD it drop the call and looks like it tag as answering machine.

[ed123]

Aug 30 08:37:17] -- <Local/55558600051@default-0000000b;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 30 08:37:18] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 30 08:37:18] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 30 08:37:20] -- SIP/VOIP-00000009 answered Local/MYPHONENUMBER@default-0000000a;2
[Aug 30 08:37:20] > Channel Local/MYPHONENUMBER@default-0000000a;1 was answered
[Aug 30 08:37:20] -- Executing [8369@default:1] Playback("Local/MYPHONENUMBER@default-0000000a;1", "sip-silence") in new stack
[Aug 30 08:37:20] -- <Local/MYPHONENUMBER@default-0000000a;1> Playing 'sip-silence.gsm' (language 'en')
[Aug 30 08:37:20] -- Executing [8369@default:2] AGI("Local/MYPHONENUMBER@default-0000000a;1", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 30 08:37:20] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=T3PH))
[Aug 30 08:37:20] -- <Local/MYPHONENUMBER@default-0000000a;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 30 08:37:20] -- Executing [8369@default:3] AMD("Local/MYPHONENUMBER@default-0000000a;1", "2000,2000,1000,5000,120,50,4,256") in new stack
[Aug 30 08:37:20] -- AMD: Local/MYPHONENUMBER@default-0000000a;1 MYPHONENUMBER (N/A) (Fmt: slin)
[Aug 30 08:37:20] -- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256] maximumWordLength [5000]
[Aug 30 08:37:20] -- AMD: Channel [Local/MYPHONENUMBER@default-0000000a;1]. Changed state to STATE_IN_SILENCE
[Aug 30 08:37:20] > 0x7fc208014a80 -- Strict RTP switching to RTP remote address 91.220.132.12:6720 as source
[Aug 30 08:37:20] -- Executing [h@default:1] AGI("Local/MYPHONENUMBER@default-0000000a;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----0") in new stack
[Aug 30 08:37:20] > 0x7fc208014a80 -- Strict RTP learning complete - Locking on source address 91.220.132.12:6720
[Aug 30 08:37:20] -- AMD: Channel [SIP/VOIP-00000009]. Word detected. iWordsCount:1
[Aug 30 08:37:20] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 180
[Aug 30 08:37:20] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 20
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Changed state to STATE_IN_SILENCE
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 60
[Aug 30 08:37:21] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Word detected. iWordsCount:2
[Aug 30 08:37:21] -- <Local/MYPHONENUMBER@default-0000000a;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --17-----0 completed, returning 0
[Aug 30 08:37:21] == Spawn extension (default, MYPHONENUMBER, 2) exited non-zero on 'Local/MYPHONENUMBER@default-0000000a;2'
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Changed state to STATE_IN_SILENCE
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 140
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Word detected. iWordsCount:3
[Aug 30 08:37:22] -- AMD: Channel [SIP/VOIP-00000009]. Changed state to STATE_IN_SILENCE
[Aug 30 08:37:22] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 500
[Aug 30 08:37:22] -- AMD: Channel [SIP/VOIP-00000009]. Word detected. iWordsCount:4
[Aug 30 08:37:22] -- AMD: Channel [SIP/VOIP-00000009]. ANSWERING MACHINE: iWordsCount:4
[Aug 30 08:37:22] -- Executing [8369@default:4] AGI("SIP/VOIP-00000009", "VD_amd.agi,8369") in new stack
[Aug 30 08:37:22] -- Launched AGI Script /usr/share/asterisk/agi-bin/VD_amd.agi
[Aug 30 08:37:22] -- <SIP/VOIP-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 30 08:37:23] -- <SIP/VOIP-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 30 08:37:23] -- <SIP/VOIP-00000009>AGI Script VD_amd.agi completed, returning 4
[Aug 30 08:37:23] == Spawn extension (default, 8369, 4) exited non-zero on 'SIP/VOIP-00000009'
[Aug 30 08:37:23] -- Executing [h@default:1] AGI("SIP/VOIP-00000009", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 30 08:37:23] -- <SIP/VOIP-00000009>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[blackbird2306]

Native answering machine detection in Asterisk is not 100 % accurate. There is often a false positive AMDSTATUS. In your first example it seems you are not talking (test call) and the AMD considers the called party as answering machine and will hangup or do whatever your "AMD send to Action" is selected in your campaign settings.
These are the default AMD settings (the values are in milliseconds):
initialSilence [2000]
Is maximum initial silence duration before greeting.
If this is exceeded, the result is detection as a MACHINE

greeting [2000]
is the maximum length of a greeting.
If this is exceeded, the result is detection as a MACHINE

afterGreetingSilence [1000]
Is the silence after detecting a greeting.
If this is exceeded, the result is detection as a HUMAN

totalAnalysisTime [5000]
Is the maximum time allowed for the algorithm
to decide on whether the audio represents a HUMAN, or a MACHINE

minimumWordLength [120]
Is the minimum duration of Voice considered to be a word

betweenWordsSilence [50]
Is the minimum duration of silence after a word to consider the audio that follows to be a new word

maximumNumberOfWords [4]
Is the maximum number of words in a greeting
If this is REACHED, then the result is detection as a MACHINE

silenceThreshold [256]
What is the average level of noise from 0 to 32767 which if not exceeded, should be considered silence

maximumWordLength [5000]
Is the maximum duration of a word to accept.
If exceeded, then the result is detection as a MACHINE

[Aug 30 07:06:27] -- AMD: Channel [SIP/VOIP-00000004]. ANSWERING MACHINE: silenceDuration:2000 initialSilence:2000
In your first example your call was detected as MACHINE, because your initial silence of 2 seconds (no talking at start) was exceeded.


[Aug 30 08:37:20] -- AMD: Channel [SIP/VOIP-00000009]. Word detected. iWordsCount:1
[Aug 30 08:37:20] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 180
[Aug 30 08:37:20] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 20
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Changed state to STATE_IN_SILENCE
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 60
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Word detected. iWordsCount:2
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Changed state to STATE_IN_SILENCE
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 140
[Aug 30 08:37:21] -- AMD: Channel [SIP/VOIP-00000009]. Word detected. iWordsCount:3
[Aug 30 08:37:22] -- AMD: Channel [SIP/VOIP-00000009]. Changed state to STATE_IN_SILENCE
[Aug 30 08:37:22] -- AMD: Channel [SIP/VOIP-00000009]. Detected Talk, previous silence duration: 500
[Aug 30 08:37:22] -- AMD: Channel [SIP/VOIP-00000009]. Word detected. iWordsCount:4
[Aug 30 08:37:22] -- AMD: Channel [SIP/VOIP-00000009]. ANSWERING MACHINE: iWordsCount:4
In your second example the AMD decides as answering machine too. For futher details look into your AGI debug log, then you will see why.

My advice: Don't use AMD it is not reliable and it adds delay of the call being sent to an agent about 3-5 seconds during analysis time.

[ed123]

Hi,

Thanks for the information but we required to atleast filter some of answering machines.

We activate it in our production and same output no calls were passed to the agent even if the call is active or live. the amd still hungup and disposed it.

[mflorell]

I know there were several bugs affecting Asterisk AMD, we submitted patches and it should work in Asterisk 13.21.0

[ed123]

Hi,

I think the problem is NAT . I tried to install goautodial 3.0 and same result. By the way we use pfsense.

logs

---
[Aug 31 06:24:06] > Channel SIP/1001-00000007 was answered.
[Aug 31 06:24:06] -- Executing [8600051@default:1] MeetMe("SIP/1001-00000007", "8600051,F") in new stack
[Aug 31 06:24:06] == Parsing '/etc/asterisk/meetme.conf': [Aug 31 06:24:06] == Found
[Aug 31 06:24:06] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Aug 31 06:24:06] == Found
[Aug 31 06:24:06] -- Created MeetMe conference 1023 for conference '8600051'
[Aug 31 06:24:06] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 31 06:24:06] -- <SIP/1001-00000007> Playing 'conf-onlyperson.gsm' (language 'en')
[Aug 31 06:24:07] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 31 06:24:07] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 31 06:24:11]
<--- SIP read from UDP:192.168.1.56:34076 --->


<------------->
[Aug 31 06:24:11] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 31 06:24:11] -- Executing [32MYPHONENUMBER@default:1] AGI("Local/32MYPHONENUMBER@default-00000008;2", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 31 06:24:11] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TEST123))
[Aug 31 06:24:11] -- <Local/32MYPHONENUMBER@default-00000008;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 31 06:24:11] -- Executing [32MYPHONENUMBER@default:2] Dial("Local/32MYPHONENUMBER@default-00000008;2", "SIP/9898MYPHONENUMBER@VOIP,60,tToR") in new stack
[Aug 31 06:24:11] == Using SIP RTP CoS mark 5
[Aug 31 06:24:11] Audio is at 17928
[Aug 31 06:24:11] Adding codec 0x4 (ulaw) to SDP
[Aug 31 06:24:11] Adding codec 0x8 (alaw) to SDP
[Aug 31 06:24:11] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 31 06:24:11] Reliably Transmitting (no NAT) to 91.220.132.12:5060:
INVITE sip:9898MYPHONENUMBER@91.220.132.12 SIP/2.0
Via: SIP/2.0/UDP MYPUBLICIP:5060;branch=z9hG4bK73a6ec1f
Max-Forwards: 70
From: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;tag=as32cb81a2
To: <sip:9898MYPHONENUMBER@91.220.132.12>
Contact: <sip:14046927964@MYPUBLICIP:5060>
Call-ID: 641cfcd07dcd54d776559c9d0392c165@MYPUBLICIP:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Date: Fri, 31 Aug 2018 10:24:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 170015533 170015533 IN IP4 MYPUBLICIP
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 MYPUBLICIP
t=0 0
m=audio 17928 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

---
[Aug 31 06:24:11] -- Called SIP/9898MYPHONENUMBER@VOIP
[Aug 31 06:24:11] Retransmitting #1 (no NAT) to 91.220.132.12:5060:
INVITE sip:9898MYPHONENUMBER@91.220.132.12 SIP/2.0
Via: SIP/2.0/UDP MYPUBLICIP:5060;branch=z9hG4bK73a6ec1f
Max-Forwards: 70
From: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;tag=as32cb81a2
To: <sip:9898MYPHONENUMBER@91.220.132.12>
Contact: <sip:14046927964@MYPUBLICIP:5060>
Call-ID: 641cfcd07dcd54d776559c9d0392c165@MYPUBLICIP:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Date: Fri, 31 Aug 2018 10:24:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 170015533 170015533 IN IP4 MYPUBLICIP
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 MYPUBLICIP
t=0 0
m=audio 17928 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

---
[Aug 31 06:24:11]
<--- SIP read from UDP:91.220.132.12:5060 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP MYPUBLICIP:5060;branch=z9hG4bK73a6ec1f
From: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;tag=as32cb81a2
Call-ID: 641cfcd07dcd54d776559c9d0392c165@MYPUBLICIP:5060
To: <sip:9898MYPHONENUMBER@91.220.132.12>
Content-Length: 0

<------------->
[Aug 31 06:24:11] --- (7 headers 0 lines) ---
[Aug 31 06:24:12]
<--- SIP read from UDP:91.220.132.12:5060 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP MYPUBLICIP:5060;branch=z9hG4bK73a6ec1f
From: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;tag=as32cb81a2
Call-ID: 641cfcd07dcd54d776559c9d0392c165@MYPUBLICIP:5060
To: <sip:9898MYPHONENUMBER@91.220.132.12>
Content-Length: 0

[ed123]

<------------->
[Aug 31 06:24:12] --- (7 headers 0 lines) ---
[Aug 31 06:24:12]
<--- SIP read from UDP:91.220.132.12:5060 --->
SIP/2.0 183 Session Progress
CSeq: 102 INVITE
Via: SIP/2.0/UDP MYPUBLICIP:5060;branch=z9hG4bK73a6ec1f
From: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;tag=as32cb81a2
Call-ID: 641cfcd07dcd54d776559c9d0392c165@MYPUBLICIP:5060
To: <sip:9898MYPHONENUMBER@91.220.132.12>;tag=112404453008105
Contact: <sip:91.220.132.12:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 209

v=0
o=- 1377272513 1377272514 IN IP4 91.220.132.12
s=VoipSIP
c=IN IP4 91.220.132.12
t=0 0
m=audio 6254 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Aug 31 06:24:12] --- (9 headers 10 lines) ---
[Aug 31 06:24:12] list_route: hop: <sip:91.220.132.12:5060;transport=udp>
[Aug 31 06:24:12] Found RTP audio format 0
[Aug 31 06:24:12] Found RTP audio format 101
[Aug 31 06:24:12] Found audio description format PCMU for ID 0
[Aug 31 06:24:12] Found audio description format telephone-event for ID 101
[Aug 31 06:24:12] Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Aug 31 06:24:12] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Aug 31 06:24:12] Peer audio RTP is at port 91.220.132.12:6254
[Aug 31 06:24:12] -- SIP/VOIP-00000008 is making progress passing it to Local/32MYPHONENUMBER@default-00000008;2
[Aug 31 06:24:16]
<--- SIP read from UDP:91.220.132.12:5060 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP MYPUBLICIP:5060;branch=z9hG4bK73a6ec1f
From: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;tag=as32cb81a2
Call-ID: 641cfcd07dcd54d776559c9d0392c165@MYPUBLICIP:5060
To: <sip:9898MYPHONENUMBER@91.220.132.12>;tag=112404453008105
Contact: <sip:91.220.132.12:5060;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER, PUBLISH, UPDATE
Content-Type: application/sdp
Content-Length: 209

v=0
o=- 1377272513 1377272514 IN IP4 91.220.132.12
s=VoipSIP
c=IN IP4 91.220.132.12
t=0 0
m=audio 6254 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Aug 31 06:24:16] --- (10 headers 10 lines) ---
[Aug 31 06:24:16] list_route: hop: <sip:91.220.132.12:5060;transport=udp>
[Aug 31 06:24:16] set_destination: Parsing <sip:91.220.132.12:5060;transport=udp> for address/port to send to
[Aug 31 06:24:16] set_destination: set destination to 91.220.132.12:5060
[Aug 31 06:24:16] Transmitting (no NAT) to 91.220.132.12:5060:
ACK sip:91.220.132.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP MYPUBLICIP:5060;branch=z9hG4bK38aa98ed
Max-Forwards: 70
From: "V8310624110000000006" <sip:14046927964@MYPUBLICIP>;tag=as32cb81a2
To: <sip:9898MYPHONENUMBER@91.220.132.12>;tag=112404453008105
Contact: <sip:14046927964@MYPUBLICIP:5060>
Call-ID: 641cfcd07dcd54d776559c9d0392c165@MYPUBLICIP:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Content-Length: 0