Help with Fax Please

I am having fax problems on my asterisk system and would appreciate any help that can be provided. I have searched several forums and have been unable to resolve my problem. Incoming faxes do not work, and outgoing faxes only sometimes work. In detail, when i send a fax to the fax machine in question, the two fax machines ring, then connect for a couple of seconds and then immediately disconnect. I have tried multiple configurations on the ATA and on Asterisk and have been unable to resolve the problem. When using a standard analog phone in place of the fax machine voice quality both directions is clear and works fine, only faxing does not work.

I am more than happy to post additional logs and perform additional tests to resolve the problem. Thanks in advance.

Asterisk 1.4.18.1
FreePBX 2.4.1.0
Aastra 9133i IP Phones
Grandstream HandyTone HT-286 ATA version 1.1.0.26 connected to HP LaserJet 3050 Analog Fax Machine
Digium TE212P PRI Card, 16 channels, digital lines provided by Cbeyond Communications

The ATA is configured with extension 2019 and DID of 2146236412. The IP address of the asterisk box is 172.31.40.10. The IP address of the ATA is 172.31.40.116.

/etc/asterisk/udptl.conf
[general]
udptlstart=4000
udptlend=4999
T38FaxUdpEC = t38UDPRedundancy
T38FaxMaxDatagram = 400
udptlfecentries = 1
udptlfecspan = 3

/etc/asterisk/sip_general_custom.conf
t38pt_udptl=yes
t38pt_rtp=no
t38pt_tcp=no

/etc/asterisk/sip_additional.conf
[2019]
type=friend
secret=******
record_out=Never
record_in=Never
qualify=yes
port=5060
pickupgroup=
nat=no
[email protected]
host=dynamic
dtmfmode=rfc2833
dial=SIP/2019
context=from-internal
canreinvite=no
callgroup=
callerid=device <2019>
accountcode=
call-limit=50

Included below is a list of settings for the Grandstream HandyTone HT-286 ATA:
[Status]
Product Model: HT287 Rev4.1
Software Version: Program-- 1.1.0.26 Bootloader-- 1.1.0.1 HTML-- 1.1.0.26 VOC-- 1.0.0.13
System Up Time: 1 day(s) 0 hour(s) 56 minute(s)
Registered: Yes
PPPoE Link Up: disabled
NAT:
NAT Mapped IP: 0.0.0.0
NAT Mapped Port: 0

[Advanced Settings 1]
Admin Password:
SIP Server: 172.31.40.10
Outbound Proxy: 172.31.40.10
SIP User ID: 2019
Authenticate ID: 2019
Authenticate Password:
Name: Fax Machine
Home NPA:
Advanced Options:
Preferred Vocoder:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: iLBC
choice 7: PCMU
G723 rate: 6.3kbps encoding rate
iLBC frame size: 20ms
iLBC payload type: 97
Silence Suppression: No
Voice Frames per TX: 2
Fax Mode: T.38 (Auto Detect)
Layer 3 QoS: 48
Layer 2 QoS: 802.1Q/VLAN Tag: 0 802.1p priority value: 0
Allow incoming SIP messages from SIP proxy only: Yes
Use DNS SRV: No
User ID is phone number: No
SIP Registration: Yes
Unregister On Reboot: No
Register Expiration: 3600
Early Dial: No
Allow outgoing call without Registration: No
Dial Plan Prefix:
No Key Entry Timeout: 4
Use # as Dial Key: No
local SIP port: 5060
local RTP port: 5004
Use random port: Yes
SIP Registration Failure Retry Wait Time: 20
NAT Traversal: No
keep-alive interval: 20
Use NAT IP:
Proxy-Require:
SUBSCRIBE for MWI: No, do not send SUBSCRIBE for Message Waiting Indication
Offhook Auto-Dial:
Enable Call Features: No
Use Bell-style 3-way Conference: No
Disable Call-Waiting: Yes
Disable Call-Waiting Caller-ID: Yes
Send DTMF: in-audio, via RTP (RFC2833), via SIP INFO
DTMF Payload Type: 101
Send Flash Event: No

[Advanced Settings 2]
Onhook Threshold: 800 ms
FXS Impedance: 600 Ohm (North America)
Caller ID Scheme: Bellcore (North America)
Onhook Voltage: 36V
Polarity Reversal: No
NTP Server: 172.31.40.10
Send Anonymous: No
Anonymous Method: Use From Header
Time to ring: 30 seconds
Special Feature: Standard
CBCOM Encode - SIP: None RT©P: None T38: None
CBCOM Encoder 1.1 Key:
Syslog Server:
Syslog Level: NONE
Session Expiration: 180
Min-SE: 90
Caller Request Timer: No
Callee Request Timer: No
Force Timer: No
UAC Specify Refresher: Omit (Recommended)
UAS Specify Refresher: UAC
Force INVITE: No (Always refresh with INVITE instead of UPDATE)

Firmware Upgrade and Provisioning: Upgrade Via TFTP
Firmware Server Path: 172.31.40.10
Configure Server Path: 172.31.40.10
Firmware File Prefix:
Firmware File Postfix:
Config File Prefix:
Config File Postfix:
Retry-after(minutes): 1
Automatic Upgrade: No
Always Check for New Firmware
Firmware Key:
Authenticate Conf File: No
Lock keypad update: No
Allow conf SIP Account in Basic Settings: No
Override MTU Size: 0
Volume Amplification: TX +6dB RX +6dB
Powerline Ring Tone:
Frequency (Hz): 20
ON (x10ms): 200
OFF (x10ms): 400

Call Progress Tones:
Frequency 1 Dial Tone: 350
Frequency 2 Dial Tone: 350
ON (x10ms) Dial Tone: 0
OFF (x10ms) Dial Tone: 0

Frequency 1 Recall Dial Tone: 350
Frequency 2 Recall Dial Tone: 440
ON (x10ms) Recall Dial Tone: 10
OFF (x10ms) Recall Dial Tone: 10

Frequency 1 Message Waiting: 350
Frequency 2 Message Waiting: 440
ON (x10ms) Message Waiting: 10
OFF (x10ms) Message Waiting: 10

Frequency 1 Confirmation: 350
Frequency 2 Confirmation: 440
ON (x10ms) Confirmation: 10
OFF (x10ms) Confirmation: 10

Frequency 1 Audible Ringing: 440
Frequency 2 Audible Ringing: 480
ON (x10ms) Audible Ringing: 200
OFF (x10ms) Audible Ringing: 400

Frequency 1 Busy Tone: 480
Frequency 2 Busy Tone: 620
ON (x10ms) Busy Tone: 50
OFF (x10ms) Busy Tone: 50

Frequency 1 Reorder Tone: 480
Frequency 2 Reorder Tone: 620
ON (x10ms) Reorder Tone: 25
OFF (x10ms) Reorder Tone: 25

Frequency 1 Receiver Offhook Tone: 1400
Frequency 2 Receiver Offhook Tone: 2600
ON (x10ms) Receiver Offhook Tone: 10
OFF (x10ms) Receiver Offhook Tone: 10

Disable Line Echo Canceller (LEC): Yes

Included below is a capture of an incoming fax attempt from asterisk cli:
[email protected]:~ $ asterisk -vvvvvvvvr
Asterisk 1.4.18.1, Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer [email protected]
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
Connected to Asterisk 1.4.18.1 currently running on voip (pid = 15048)
Verbosity is at least 9
voip*CLI>sip set debug peer 2019

SIP Debugging Enabled for IP: 172.31.40.116:33856
Reliably Transmitting (no NAT) to 172.31.40.116:33856:
OPTIONS sip:[email protected]:33856 SIP/2.0

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK5d67a831;rport

From: “Unknown” sip:[email protected];tag=as3c20be52

To: sip:[email protected]:33856

Contact: sip:[email protected]

Call-ID: [email protected]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 13 Aug 2008 21:44:09 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0


<— SIP read from 172.31.40.116:33856 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK5d67a831;rport

From: “Unknown” sip:[email protected];tag=as3c20be52

To: sip:[email protected]:33856;tag=1d045d05bd04fee3

Call-ID: [email protected]

CSeq: 102 OPTIONS

User-Agent: Grandstream HT287 1.1.0.26

Contact: sip:[email protected]:33856

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Supported: replaces, timer

Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Reliably Transmitting (no NAT) to 172.31.40.116:33856:
OPTIONS sip:[email protected]:33856 SIP/2.0

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK187bf69a;rport

From: “Unknown” sip:[email protected];tag=as6a2dac6f

To: sip:[email protected]:33856

Contact: sip:[email protected]40.10

Call-ID: [email protected]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 13 Aug 2008 21:45:09 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0


<— SIP read from 172.31.40.116:33856 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK187bf69a;rport

From: “Unknown” sip:[email protected];tag=as6a2dac6f

To: sip:[email protected]:33856;tag=e93b777f2ceb6adb

Call-ID: [email protected]

CSeq: 102 OPTIONS

User-Agent: Grandstream HT287 1.1.0.26

Contact: sip:[email protected]:33856

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Supported: replaces, timer

Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
– Accepting call from ‘6194564543’ to ‘2146236412’ on channel 0/1, span 1
– Executing [[email protected]:1] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mDID=2146236412e[0;37;40m”) in new stack
– Executing [[email protected]:2] e[1;36;40mGotoe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40ms|1e[0;37;40m”) in new stack
– Goto (from-zaptel,s,1)
– Executing [[email protected]:1] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mEntering from-zaptel with DID == 2146236412e[0;37;40m”) in new stack
– Executing [[email protected]:2] e[1;36;40mRinginge[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack
– Executing [[email protected]:3] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mDID=2146236412e[0;37;40m”) in new stack
– Executing [[email protected]:4] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mDID is now 2146236412e[0;37;40m”) in new stack
– Executing [[email protected]:5] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m1?zapok:notzape[0;37;40m”) in new stack
– Goto (from-zaptel,s,8)
– Executing [[email protected]:8] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mIs a Zaptel Channele[0;37;40m”) in new stack
– Executing [[email protected]:9] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mCHAN=1-1e[0;37;40m”) in new stack
– Executing [[email protected]:10] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mCHAN=1e[0;37;40m”) in new stack
– Executing [[email protected]:11] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mfrom-zaptel-1|2146236412|1e[0;37;40m”) in new stack
– Executing [[email protected]:12] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mReturned from Macro from-zaptel-1e[0;37;40m”) in new stack
– Executing [[email protected]:13] e[1;36;40mGotoe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mfrom-pstn|2146236412|1e[0;37;40m”) in new stack
– Goto (from-pstn,2146236412,1)
– Executing [[email protected]:1] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m__FROM_DID=2146236412e[0;37;40m”) in new stack
– Executing [[email protected]:2] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m1 ?cidoke[0;37;40m”) in new stack
– Goto (from-pstn,2146236412,4)
– Executing [[email protected]:4] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mCallerID is “TRACE SECURITY” <6194564543>e[0;37;40m”) in new stack
– Executing [[email protected]:5] e[1;36;40mSetMusicOnHolde[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mnonee[0;37;40m”) in new stack
– Executing [[email protected]:6] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m__MOHCLASS=nonee[0;37;40m”) in new stack
– Executing [[email protected]:7] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mFAX_RX=disablede[0;37;40m”) in new stack
– Executing [[email protected]:8] e[1;36;40mGotoe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mfrom-did-direct|2019|1e[0;37;40m”) in new stack
– Goto (from-did-direct,2019,1)
– Executing [[email protected]:1] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mexten-vm|novm|2019e[0;37;40m”) in new stack
– Executing [[email protected]:1] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40muser-calleride[0;37;40m”) in new stack
– Executing [[email protected]:1] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40muser-callerid: TRACE SECURITY 6194564543e[0;37;40m”) in new stack
– Executing [[email protected]:2] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mAMPUSER=6194564543e[0;37;40m”) in new stack
– Executing [[email protected]:3] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m0?reporte[0;37;40m”) in new stack
– Executing [[email protected]:4] e[1;36;40mExecIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m1|Set|REALCALLERIDNUM=6194564543e[0;37;40m”) in new stack
– Executing [[email protected]:5] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mREALCALLERIDNUM is 6194564543e[0;37;40m”) in new stack
– Executing [[email protected]:6] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mAMPUSER=e[0;37;40m”) in new stack
– Executing [[email protected]:7] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mAMPUSERCIDNAME=e[0;37;40m”) in new stack
– Executing [[email protected]:8] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m1?reporte[0;37;40m”) in new stack
– Goto (macro-user-callerid,s,13)
– Executing [[email protected]:13] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mTTL: ARG1: novme[0;37;40m”) in new stack
– Executing [[email protected]:14] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m0?continuee[0;37;40m”) in new stack
– Executing [[email protected]:15] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m__TTL=64e[0;37;40m”) in new stack
– Executing [[email protected]:16] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m1?continuee[0;37;40m”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [[email protected]:23] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mUsing CallerID “TRACE SECURITY” <6194564543>e[0;37;40m”) in new stack
– Executing [[email protected]:2] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mRingGroupMethod=nonee[0;37;40m”) in new stack
– Executing [[email protected]:3] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mVMBOX=novme[0;37;40m”) in new stack
– Executing [[email protected]:4] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mEXTTOCALL=2019e[0;37;40m”) in new stack
– Executing [[email protected]:5] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mCFUEXT=e[0;37;40m”) in new stack
– Executing [[email protected]:6] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mCFBEXT=e[0;37;40m”) in new stack
– Executing [[email protected]:7] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mRT=”“e[0;37;40m”) in new stack
– Executing [[email protected]:8] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mrecord-enable|2019|INe[0;37;40m”) in new stack
– Executing [[email protected]:1] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m0?2:4e[0;37;40m”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] e[1;36;40mAGIe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mrecordingcheck|20080813-164522|1218663922.4991e[0;37;40m”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080813-164522|1218663922.4991: Inbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mNo recording needede[0;37;40m”) in new stack
– Executing [[email protected]:9] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mdial||trwW|2019e[0;37;40m”) in new stack
– Executing [[email protected]:1] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m0?diale[0;37;40m”) in new stack
– Executing [[email protected]:2] e[1;36;40mSetMusicOnHolde[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mnonee[0;37;40m”) in new stack
– Executing [[email protected]:3] e[1;36;40mAGIe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mdialparties.agie[0;37;40m”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
dialparties.agi: Caller ID name is ‘TRACE SECURITY’ number is '6194564543’
dialparties.agi: USE_CONFIRMATION: 'FALSE’
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 2019 to extension map
– dialparties.agi: Extension 2019 cf is disabled
– dialparties.agi: Extension 2019 do not disturb is disabled
> dialparties.agi: extnum 2019 has: cw: 0; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 2019 has ExtensionState: 0
– dialparties.agi: Checking CW and CFB status for extension 2019
– dialparties.agi: dbset CALLTRACE/2019 to 6194564543
– dialparties.agi: Filtered ARG3: 2019
== Manager ‘admin’ logged off from 127.0.0.1
– AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] e[1;36;40mDiale[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mSIP/2019||trwWe[0;37;40m”) in new stack
Audio is at 172.31.40.10 port 19472
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 172.31.40.116:33856:
INVITE sip:[email protected]:33856 SIP/2.0

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK474fac47;rport

From: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

To: sip:[email protected]:33856

Contact: sip:[email protected]

Call-ID: [email protected]

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 13 Aug 2008 21:45:22 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Type: application/sdp

Content-Length: 264

v=0

o=root 15048 15048 IN IP4 172.31.40.10

s=session

c=IN IP4 172.31.40.10

t=0 0

m=audio 19472 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=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


-- Called 2019

<— SIP read from 172.31.40.116:33856 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK474fac47;rport

From: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

To: sip:[email protected]:33856

Call-ID: [email protected]

CSeq: 102 INVITE

User-Agent: Grandstream HT287 1.1.0.26

Content-Length: 0

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

<— SIP read from 172.31.40.116:33856 —>
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK474fac47;rport

From: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

To: sip:[email protected]:33856;tag=f04ed39e4256530f

Call-ID: [email protected]

CSeq: 102 INVITE

User-Agent: Grandstream HT287 1.1.0.26

Content-Length: 0

<------------->
— (8 headers 0 lines) —
– SIP/2019-b7d3dcc8 is ringing

<— SIP read from 172.31.40.116:33856 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK474fac47;rport

From: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

To: sip:[email protected]:33856;tag=f04ed39e4256530f

Call-ID: [email protected]

CSeq: 102 INVITE

User-Agent: Grandstream HT287 1.1.0.26

Contact: sip:[email protected]:33856

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Type: application/sdp

Supported: replaces, timer

Content-Length: 214

v=0

o=2019 8000 8000 IN IP4 172.31.40.116

s=SIP Call

c=IN IP4 172.31.40.116

t=0 0

m=audio 61716 RTP/AVP 0 101

a=sendrecv

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

<------------->
— (12 headers 11 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.31.40.116:61716
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.31.40.116:61716
list_route: hop: sip:[email protected]:33856
set_destination: Parsing sip:[email protected]:33856 for address/port to send to
set_destination: set destination to 172.31.40.116, port 33856
Transmitting (no NAT) to 172.31.40.116:33856:
ACK sip:[email protected]:33856 SIP/2.0

Via: SIP/2.0/UDP 172.31.40.10:5060;branch=z9hG4bK1431bdc9;rport

From: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

To: sip:[email protected]:33856;tag=f04ed39e4256530f

Contact: sip:[email protected]

Call-ID: [email protected]

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0


-- SIP/2019-b7d3dcc8 answered Zap/1-1

<— SIP read from 172.31.40.116:33856 —>
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK609649777ed0b241

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Contact: sip:[email protected]:33856

Supported: replaces, timer

Call-ID: [email protected]

CSeq: 7169 INVITE

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Type: application/sdp

Content-Length: 351

v=0

o=2019 8000 8001 IN IP4 172.31.40.116

s=SIP Call

c=IN IP4 172.31.40.116

t=0 0

m=image 61716 udptl t38

a=T38FaxVersion:0

a=T38MaxBitRate:9600

a=T38FaxFillBitRemoval:0

a=T38FaxTranscodingMMR:0

a=T38FaxTranscodingJBIG:0

a=T38FaxRateManagement:transferredTCF

a=T38FaxMaxBuffer:400

a=T38FaxMaxDatagram:280

a=T38FaxUdpEC:t38UDPRedundancy

<------------->
— (13 headers 15 lines) —
Sending to 172.31.40.116 : 33856 (no NAT)
Got T.38 offer in SDP in dialog [email protected]
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid [email protected]
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)

<— Transmitting (no NAT) to 172.31.40.116:33856 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK609649777ed0b241;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7169 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

Content-Length: 0

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

<— Reliably Transmitting (no NAT) to 172.31.40.116:33856 —>
SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK609649777ed0b241;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7169 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

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

<— SIP read from 172.31.40.116:33856 —>
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK609649777ed0b241

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Contact: sip:[email protected]:33856

Call-ID: [email protected]

CSeq: 7169 ACK

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Length: 0

<------------->
— (11 headers 0 lines) —

<— SIP read from 172.31.40.116:33856 —>
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK8d26f2fd158ca38d

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Contact: sip:[email protected]:33856

Supported: replaces, timer

Call-ID: [email protected]

CSeq: 7170 INVITE

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Type: application/sdp

Content-Length: 214

v=0

o=2019 8000 8002 IN IP4 172.31.40.116

s=SIP Call

c=IN IP4 172.31.40.116

t=0 0

m=audio 61716 RTP/AVP 0 101

a=sendrecv

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

<------------->
— (13 headers 11 lines) —
Sending to 172.31.40.116 : 33856 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.31.40.116:61716
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.31.40.116:61716

<— Transmitting (no NAT) to 172.31.40.116:33856 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK8d26f2fd158ca38d;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7170 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

Content-Length: 0

<------------>
Audio is at 172.31.40.10 port 19472
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 172.31.40.116:33856 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK8d26f2fd158ca38d;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7170 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

Content-Type: application/sdp

Content-Length: 240

v=0

o=root 15048 15049 IN IP4 172.31.40.10

s=session

c=IN IP4 172.31.40.10

t=0 0

m=audio 19472 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

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

<— SIP read from 172.31.40.116:33856 —>
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKf1652bcbc94c2e23

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Contact: sip:[email protected]:33856

Call-ID: [email protected]

CSeq: 7170 ACK

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Length: 0

<------------->
— (11 headers 0 lines) —

<— SIP read from 172.31.40.116:33856 —>
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKfa0541f3eb83ef44

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Contact: sip:[email protected]:33856

Supported: replaces, timer

Call-ID: [email protected]

CSeq: 7171 INVITE

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Type: application/sdp

Content-Length: 351

v=0

o=2019 8000 8003 IN IP4 172.31.40.116

s=SIP Call

c=IN IP4 172.31.40.116

t=0 0

m=image 61716 udptl t38

a=T38FaxVersion:0

a=T38MaxBitRate:9600

a=T38FaxFillBitRemoval:0

a=T38FaxTranscodingMMR:0

a=T38FaxTranscodingJBIG:0

a=T38FaxRateManagement:transferredTCF

a=T38FaxMaxBuffer:400

a=T38FaxMaxDatagram:280

a=T38FaxUdpEC:t38UDPRedundancy

<------------->
— (13 headers 15 lines) —
Sending to 172.31.40.116 : 33856 (no NAT)
Got T.38 offer in SDP in dialog [email protected]
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid [email protected]
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)

<— Transmitting (no NAT) to 172.31.40.116:33856 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKfa0541f3eb83ef44;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7171 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

Content-Length: 0

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

<— Reliably Transmitting (no NAT) to 172.31.40.116:33856 —>
SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKfa0541f3eb83ef44;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7171 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

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

<— SIP read from 172.31.40.116:33856 —>
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK329f66ff619a7e8a

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Contact: sip:[email protected]1.40.116:33856

Call-ID: [email protected]

CSeq: 7170 ACK

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Length: 0

<------------->
— (11 headers 0 lines) —

<— SIP read from 172.31.40.116:33856 —>
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK5515260b8478321b

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Contact: sip:[email protected]:33856

Supported: replaces, timer

Call-ID: [email protected]

CSeq: 7172 INVITE

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Type: application/sdp

Content-Length: 214

v=0

o=2019 8000 8004 IN IP4 172.31.40.116

s=SIP Call

c=IN IP4 172.31.40.116

t=0 0

m=audio 61716 RTP/AVP 0 101

a=sendrecv

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

<------------->
— (13 headers 11 lines) —

<— Transmitting (no NAT) to 172.31.40.116:33856 —>
SIP/2.0 491 Request Pending

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK5515260b8478321b;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7172 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

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

<— SIP read from 172.31.40.116:33856 —>
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bK5515260b8478321b

From: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Contact: sip:[email protected]:33856

Call-ID: [email protected]

CSeq: 7172 ACK

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Length: 0

<------------->
— (11 headers 0 lines) —

<— SIP read from 172.31.40.116:33856 —>
BYE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKdda936b9b1b186c1

From: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Supported: replaces, timer

Call-ID: [email protected]

CSeq: 7173 BYE

User-Agent: Grandstream HT287 1.1.0.26

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE

Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 172.31.40.116 : 33856 (no NAT)

<— Transmitting (no NAT) to 172.31.40.116:33856 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKdda936b9b1b186c1;received=172.31.40.116

From: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7173 BYE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

Content-Length: 0

<------------>
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘Zap/1-1’ in macro ‘dial’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘Zap/1-1’ in macro ‘exten-vm’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘Zap/1-1’
– Executing [[email protected]:1] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mhangupcalle[0;37;40m”) in new stack
– Executing [[email protected]:1] e[1;36;40mResetCDRe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40mwe[0;37;40m”) in new stack
– Executing [[email protected]:2] e[1;36;40mNoCDRe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack
– Executing [[email protected]:3] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m1?skiprge[0;37;40m”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m1?skipblkvme[0;37;40m”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40m1?theende[0;37;40m”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] e[1;36;40mHangupe[0;37;40m(“e[1;35;40mZap/1-1e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘Zap/1-1’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘Zap/1-1’
– Hungup ‘Zap/1-1’


voip*CLI>
Retransmitting #1 (no NAT) to 172.31.40.116:33856:
SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKfa0541f3eb83ef44;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7171 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16


voip*CLI>
Retransmitting #2 (no NAT) to 172.31.40.116:33856:
SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKfa0541f3eb83ef44;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7171 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16


voip*CLI>
Retransmitting #3 (no NAT) to 172.31.40.116:33856:
SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKfa0541f3eb83ef44;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7171 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16


voip*CLI>
Retransmitting #4 (no NAT) to 172.31.40.116:33856:
SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 172.31.40.116:33856;branch=z9hG4bKfa0541f3eb83ef44;received=172.31.40.116

From: sip:[email protected]:33856;tag=f04ed39e4256530f

To: “TRACE SECURITY” sip:[email protected];tag=as40608fc5

Call-ID: [email protected]

CSeq: 7171 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16


voip*CLI> quit

Executing last minute cleanups

Send the call on context = ext-fax not in context=from-internal & then test.