Asterisk – installation and dial plans for WebRTC supported PJSIP clients

Asterisk is a framework or toolkit designed for VOIP systems . It can support Enterprise communication systems like PBXs, call distributors, VoIP gateways , conference bridges etc . It is open source and free to use . It is developed in C and runs in linux .

Technically , Asterisk has protocol support for many telephony technologies and protocols such as SIP , H323 . It can connect old PSTN or copper line and VOIP .

Asterisk is a framework for building multi-protocol, real-time communications applications and solutions. Asterisk is to realtime voice and video applications as what Apache is to web applications

– asterisk.org

Combine the SIP channel, the PSTN interface channel and some Dialplan script and you have a gateway.
Change the Dialplan to drop calls into a ConfBridge session and you have a conference server.
Alter it once more to route calls into voice mailboxes and you have a voicemail server.
Tie it all together and you have an amazingly powerful phone system.

– asterisk.org

A repo containing Asterisk configuration and common applications can be found at https://github.com/altanai/asteriskexamples

Asterisk as a Central signalling SIP application Server in VoIP Platform

Due to the wide array of call flow processing and media, channel, bridge management module support of asterisk, it is ideal to sit at the core of a VoIP of Communication platform solution. A WebRTC capable CPaaS overall architecture with inbound and outbound Kamailio proxy and central asterisk signaller and integration to telecom service providers over SIP trunks can be described as below

Quick Installation of Asterisk

goto /usr/src and download the preferred the version of asterisk code from
http://downloads.asterisk.org/pub/telephony/asterisk/

I am using the latest release candidate at the time writing this article asterisk-16.2.0-rc1-patch.tar.gz

Some external Dependencies apt-get install subversion

Then install the source dependencies

 sudo su
contrib/scripts/get_mp3_source.sh

This will install mp3 related programs such as


A addons/mp3
A addons/mp3/decode_ntom.c
A addons/mp3/interface.c
A addons/mp3/MPGLIB_README
A addons/mp3/common.c
A addons/mp3/huffman.h
A addons/mp3/tabinit.c
A addons/mp3/Makefile
A addons/mp3/README
A addons/mp3/decode_i386.c
A addons/mp3/dct64_i386.c
A addons/mp3/MPGLIB_TODO
A addons/mp3/mpg123.h
A addons/mp3/layer3.c
A addons/mp3/mpglib.h
Exported revision 202.

Actual dependencies will be installed via install_prereq script

contrib/scripts/install_prereq install

Output snippet

Run configure which will create scripts for next processes

 ./configure

Build third party scripts

make -j2

After build , to run the installation

make install

Asterisk PBX setup

make basic-pbx

The output should be

Installing basic-pbx config files…
Installing file configs/basic-pbx/README
Installing file configs/basic-pbx/asterisk.conf
Installing file configs/basic-pbx/cdr.conf
Installing file configs/basic-pbx/cdr_custom.conf
Installing file configs/basic-pbx/confbridge.conf
Installing file configs/basic-pbx/extensions.conf
Installing file configs/basic-pbx/indications.conf
Installing file configs/basic-pbx/logger.conf
Installing file configs/basic-pbx/modules.conf
Installing file configs/basic-pbx/musiconhold.conf
Installing file configs/basic-pbx/pjsip.conf
Installing file configs/basic-pbx/voicemail.conf
Updating asterisk.conf

Also run make config to make pbx configs

make config 

start asterisk

systemctl start asterisk

connect to asterisk tool for cli

asterisk -vvvr
Asterisk 16.2.0-rc1, Copyright (C) 1999 - 2018, 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 16.2.0-rc1 currently running on ip-172-31-45-26 (pid = 13388)
ip-172-31-45-26*CLI> 

Register sip phones with asterisk PBX and make / receive calls

To make calls among users, we need to configure channel driver with sip support . Using the sip protcol the phones within the enterprise will be able to send call signals out to one another. Open pjsip.conf

Take any endpoint from template , such as

;================================
;Laverne Roberts
;Software Engineer
[1113 (endpoint-internal-d70)
auth = 1113
aors = 1113
callerid = Laverne Roberts <1113>
[1113 (auth-userpass)
password = xxxxxxxx
username = xxxxxxxx
[1113 (aor-single-reg)
mailboxes = 1113@example

and set the values in sip softphone like zoiper , register with provided creds

registering sip phone zoiper with newly created asterisk PBX

If the registration creds used are not matching with the ones defines in pjsip.conf then REGISTER request failed message is displayed

Request 'REGISTER' from '' failed for 'x.x.x.x:18475' (callid: hp8iN6oWLRdER4zvEBdiUg..) - No matching endpoint found

On correct creds used the server prints traces such as

-- Added contact 'sip:1113@x.x.x.x:44312;transport=UDP;rinstance=b8aceff08623b51e' to AOR '1113' with expiration of 60 seconds
== Endpoint 1113 is now Reachable
-- Removed contact 'sip:1113@x.x.x.x:44312;transport=UDP;rinstance=b8aceff08623b51e' from AOR '1113' due to request
== Contact 1113/sip:1113@x.x.x.x:44312;transport=UDP;rinstance=b8aceff08623b51e has been deleted
== Endpoint 1113 is now Unreachable
-- Added contact 'sip:1113@x.x.x.x:18475;transport=UDP;rinstance=5af431512ae0af3a' to AOR '1113' with expiration of 60 seconds
== Endpoint 1113 is now Reachable

Alternatively one can also create new sip endpoints

Dialplan Applications

Playback Hello World

[internal_users]
exten => 6000,1,Answer()
exten => 6000,2,Verbose("---------------- Tring Tring -------")
exten => 6000,3,Wait(1)
exten => 6000,n,Playback(silence/1&hello-world)
exten => 6000,n,Hangup()

NoOP and printing channel variables

exten=>6124,1,Verbose(2,The channel name is ${CHANNEL})
same => n,Verbose(2,The unique id is ${UNIQUEID})
same => n,Verbose(2,The caller id is ${CALLERID(all)})
same => n,Verbose(2,The datetime is ${DATETIME})
same => n,Verbose(2,The timestamp is ${TIMESTAMP})
same => n,Verbose(2,The context is ${CONTEXT})
same => n,Verbose(2,The SYSTEMNAME is ${SYSTEMNAME})
same => n,Verbose(2,The PRIORITY is ${PRIORITY})
same => n,Verbose(2,The CHANNEL is ${CHANNEL})

settings varaibles and Say

exten=>6009,1,Verbose("------------ Set variable -------")
same => n,Set(COUNT=3)
same => n,SayNumber(${COUNT})
same => n,Set(${COUNT}=10)
same => n,SayNumber(${COUNT}) 

Simple Voicemail ( also need configuration on voicemail.conf)

exten => 1235,1,VoiceMail(1235,u)               

PBX and applications

PBX cores settings 
Version:                     16.2.0-rc1
   Build Options:               BUILD_NATIVE, OPTIONAL_API
   Maximum calls:               Not set
   Maximum open file handles:   1024
   Root console verbosity:      5
   Current console verbosity:   5
   Debug level:                 0
   Maximum load average:        0.000000
   Minimum free memory:         0 MB
   Startup time:                10:27:35
   Last reload time:            10:27:35
   System:                      Linux/4.15.0-1021-aws built by root on x86_64 2019-02-11 11:48:29 UTC
   System name:                 
   Entity ID:                   0e:28:c0:44:39:5e
   PBX UUID:                    a2df96bb-6d1a-4f64-a953-cf02030e9851
   Default language:            en
   Language prefix:             Enabled
   User name and group:         /
   Executable includes:         Disabled
   Transcode via SLIN:          Enabled
   Transmit silence during rec: Disabled
   Generic PLC:                 Disabled
   Generic PLC on equal codecs: Disabled
   Min DTMF duration::          80
   Cache media frames:          Enabled
   RTP use dynamic payloads:    1
   RTP dynamic payload types:   35-63,96-127

Subsystems
   Manager (AMI):               Disabled
   Web Manager (AMI/HTTP):      Disabled
   Call data records:           Enabled
   Realtime Architecture (ARA): Disabled

 Directories
   Configuration file:          /etc/asterisk/asterisk.conf
   Configuration directory:     /etc/asterisk
   Module directory:            /usr/lib/asterisk/modules
   Spool directory:             /var/spool/asterisk
   Log directory:               /var/log/asterisk
   Run/Sockets directory:       /var/run/asterisk
   PID file:                    /var/run/asterisk/asterisk.pid
   VarLib directory:            /var/lib/asterisk
   Data directory:              /var/lib/asterisk
   ASTDB:                       /var/lib/asterisk/astdb
   IAX2 Keys directory:         /var/lib/asterisk/keys
   AGI Scripts directory:       /var/lib/asterisk/agi-bin

Asterisk Applications

Some of the application are listed below . Note that this is not an extensive list as more application are added and old ones are removed every minor release .

AddQueueMember: Dynamically adds queue members.
AlarmReceiver: Provide support for receiving alarm reports from a burglar or fire alarm panel.
AMD: Attempt to detect answering machines.
Answer: Answer a channel if ringing.
AttendedTransfer: Attended transfer to the extension provided and TRANSFER_CONTEXT
BackGround: Play an audio file while waiting for digits of an extension to go to.
BackgroundDetect: Background a file with talk detect.
BlindTransfer: Blind transfer channel(s) to the extension and context provided
Bridge: Bridge two channels.
BridgeAdd: Join a bridge that contains the specified channel.
BridgeWait: Put a call into the holding bridge.
Busy: Indicate the Busy condition.
ChanSpy: Listen to a channel, and optionally whisper into it.
ConfBridge: Conference bridge application.
Dial: Attempt to connect to another device or endpoint and bridge the call.
Dictate: Virtual Dictation Machine.
Echo: Echo media, DTMF back to the calling party
ExtenSpy: Listen to a channel, and optionally whisper into it.
ExternalIVR: Interfaces with an external IVR application.
FollowMe: Find-Me/Follow-Me application.
ForkCDR: Forks the current Call Data Record for this channel.
Hangup: Hang up the calling channel.
Log: Send arbitrary text to a selected log level.
MailboxExists: Check to see if Voicemail mailbox exists.
Milliwatt: Generate a Constant 1004Hz tone at 0dbm (mu-law).
MixMonitor: Record a call and mix the audio during the recording. Use of StopMixMonitor is required to guarantee the audio file is available for processing during dialplan execution.
Monitor: Monitor a channel.
Morsecode: Plays morse code.
MusicOnHold: Play Music On Hold indefinitely.
NoCDR: Tell Asterisk to not maintain a CDR for this channel.
NoOp: Do Nothing (No Operation).
Originate: Originate a call.
Park: Park yourself.
PauseMonitor: Pause monitoring of a channel.
PauseQueueMember: Pauses a queue member.
Pickup: Directed extension call pickup.
Playback: Play a file.
PlayTones: Play a tone list.
PrivacyManager: Require phone number to be entered, if no CallerID sent
Proceeding: Indicate proceeding.
Progress: Indicate progress.
Queue: Queue a call for a call queue.
Read: Read a variable.
SendImage: Sends an image file.
SendText: Send a Text Message on a channel.
SendURL: Send a URL.
Set: Set channel variable or function value.
SetAMAFlags: Set the AMA Flags.
SMS: Communicates with SMS service centres and SMS capable analogue phones.
SoftHangup: Hangs up the requested channel.
SpeechActivateGrammar: Activate a grammar.
SpeechBackground: Play a sound file and wait for speech to be recognized.
SpeechCreate: Create a Speech Structure.
SpeechDeactivateGrammar: Deactivate a grammar.
SpeechDestroy: End speech recognition.
SpeechLoadGrammar: Load a grammar.
SpeechProcessingSound: Change background processing sound.
SpeechStart: Start recognizing voice in the audio stream.
SpeechUnloadGrammar: Unload a grammar.
StackPop: Remove one address from gosub stack.
StartMusicOnHold: Play Music On Hold.
Stasis: Invoke an external Stasis application.
StopMixMonitor: Stop recording a call through MixMonitor, and free the recording’s file handle.
StopMonitor: Stop monitoring a channel.
StopMusicOnHold: Stop playing Music On Hold.
StopPlayTones: Stop playing a tone list.
StreamEcho: Echo media, up to ‘N’ streams of a type, and DTMF back to the calling party
VMSayName: Play the name of a voicemail user
VoiceMail: Leave a Voicemail message.
Record: Record to a file.
RemoveQueueMember: Dynamically removes queue members.
ResetCDR: Resets the Call Data Record.
RetryDial: Place a call, retrying on failure allowing an optional exit extension.
Return: Return from gosub routine.
Ringing: Indicate ringing tone.
SayNumber: Say Number.
SMS: Communicates with SMS service centres and SMS capable analogue phones.
SoftHangup: Hangs up the requested channel.
SpeechActivateGrammar: Activate a grammar.
SpeechBackground: Play a sound file and wait for speech to be recognized.
SpeechCreate: Create a Speech Structure.
SpeechDeactivateGrammar: Deactivate a grammar.
SpeechDestroy: End speech recognition.
SpeechLoadGrammar: Load a grammar.
SpeechProcessingSound: Change background processing sound.
SpeechStart: Start recognizing voice in the audio stream.
StartMusicOnHold: Play Music On Hold.
Stasis: Invoke an external Stasis application.
StopMixMonitor: Stop recording a call through MixMonitor, and free the recording’s file handle.
StopMonitor: Stop monitoring a channel.
StopMusicOnHold: Stop playing Music On Hold.
StopPlayTones: Stop playing a tone list.
StreamEcho: Echo media, up to ‘N’ streams of a type, and DTMF back to the calling party
System: Execute a system command.
VoiceMail: Leave a Voicemail message.
VoiceMailMain: Check Voicemail messages.
VoiceMailPlayMsg: Play a single voice mail msg from a mailbox by msg id.
Wait: Waits for some time.
Zapateller: Block telemarketers with SIT.

Webrtc support for asterisk over res_pjsip

Refernce project https://github.com/altanai/asteriskexamples/tree/master/webrtc_pjsip

To connect video based webrtc endpoints ensure you load the codecs and also libsrtp . Overwrite the selective conf in this folders with the existing conf of asterisk to run a basic webrtc video call . These were tested with jssip on asterisk v17 with res_pjsip.

confirm using pjsip – since chan_sip is depriciate. Confirm that yo are not using chan_sip and instead ensure using re_pjsip

module unload chan_sip
module show like res_pjsip

If pjssip is not found load it wither form menuselect or by using cli load module command .

pjssip show endpoints

*CLI> pjsip show endpoints

 Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <Identify/Endpoint.........................................................>
        Match:  <criteria.........................>
==========================================================================================

 Endpoint:  7000                                                 Invalid       0 of inf
     InAuth:  7000/7000
        Aor:  7000                                               1

 Endpoint:  8000                                                 Invalid       0 of inf
     InAuth:  8000/8000
        Aor:  8000                                               1

Generating self-signed certs

use the “ast_tls_cert” script in the “contrib/scripts” Asterisk source directory to make a self-signed certificate authority and an Asterisk certificate.

sh ast_tls_cert.sh -C localhost -O "altanai" -d .

after creating the self signed keys start the server

asterisk -vvvvvvc

channels

Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry     Peer      
10.10.10.10     1060             e8ae107f-ce90-2  (ulaw)           No       Rx: ACK                    1060 

Codecs – check for the webrtc supported audio and video codesc in the list , if not found install the modules and reload.

  31 image png          png              (PNG Image)
   6 audio g726         g726             (G.726 RFC3551)
   4 audio alaw         alaw             (G.711 a-law)
   2 audio g723         g723             (G.723.1)
  20 audio speex        speex            (SpeeX)
  21 audio speex        speex16          (SpeeX 16khz)
  22 audio speex        speex32          (SpeeX 32khz)
  24 audio g722         g722             (G722)
  25 audio siren7       siren7           (ITU G.722.1 (Siren7, licensed from Polycom))
  32 video h261         h261             (H.261 video)
  33 video h263         h263             (H.263 video)
   8 audio adpcm        adpcm            (Dialogic ADPCM)
  36 video h265         h265             (H.265 video)
  44 audio silk         silk8            (SILK Codec (8 KHz))
  45 audio silk         silk12           (SILK Codec (12 KHz))
  46 audio silk         silk16           (SILK Codec (16 KHz))
  47 audio silk         silk24           (SILK Codec (24 KHz))
  28 audio g719         g719             (ITU G.719)
  34 video h263p        h263p            (H.263+ video)
  35 video h264         h264             (H.264 video)
  19 audio g729         g729             (G.729A)
   9 audio slin         slin             (16 bit Signed Linear PCM)
  10 audio slin         slin12           (16 bit Signed Linear PCM (12kHz))
  11 audio slin         slin16           (16 bit Signed Linear PCM (16kHz))
  12 audio slin         slin24           (16 bit Signed Linear PCM (24kHz))
  13 audio slin         slin32           (16 bit Signed Linear PCM (32kHz))
  14 audio slin         slin44           (16 bit Signed Linear PCM (44kHz))
  15 audio slin         slin48           (16 bit Signed Linear PCM (48kHz))
  16 audio slin         slin96           (16 bit Signed Linear PCM (96kHz))
  17 audio slin         slin192          (16 bit Signed Linear PCM (192kHz))
   3 audio ulaw         ulaw             (G.711 u-law)
  18 audio lpc10        lpc10            (LPC10)
  27 audio testlaw      testlaw          (G.711 test-law)
  43 audio none         none             (<Null> codec)
  42 image t38          t38              (T.38 UDPTL Fax)
  39 video vp9          vp9              (VP9 video)
  38 video vp8          vp8              (VP8 video)
   5 audio gsm          gsm              (GSM)
  37 video mpeg4        mpeg4            (MPEG4 video)
  23 audio ilbc         ilbc             (iLBC)
  40 text  red          red              (T.140 Realtime Text with redundancy)
  41 text  t140         t140             (Passthrough T.140 Realtime Text)
  29 audio opus         opus             (Opus Codec)
  30 image jpeg         jpeg             (JPEG image)
   7 audio g726aal2     g726aal2         (G.726 AAL2)
   1 audio codec2       codec2           (Codec 2)
  26 audio siren14      siren14          (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))

Webrtc clients

Here is a succesful run using PJSIP as the webrtc client to communicated to another pjsip client via Asterisk server .

Note that all signalling and media isi getting proxied via the Asterisk server signalling and media plane which is in contrast to the peer to peer nature of WebRTC

Register with SIP Registrar on Asterisk
Specify the contact URI as the Asterisk server too
Incoming Call
WebRTC Call in progress via asterisk SIP and Media Server
Signalling and Media Proxy Via Asterisk

Debugging

set debug all

pjsip set logger on
rtp set debug on

setting debugger on a peer

SIP SET DEBUG PEER 1061

Chrome WebRTC Internals

References :


Freeswitch PBX system

This article talks about setting up an in-house hosted Enterprise PBX system for sure and private communication within enterprise communication.

FreeSWITCH is free and open source communications software licensed under Mozilla Public License. It if often the core of voice core to provider call routing and media control . Its core library, libfreeswitch, is capable of being embedded into other projects, as well as being used as a stand-alone application. Read more about FreeSwitch SIP and Media Server

Installation of Freeswitch on hosted server

Fetching source code

apt-get install git
git clone https://stash.freeswitch.org/scm/fs/freeswitch.git

Verify installation by checking version

freeswitch -version
FreeSWITCH version: 1.9.0-742-8f1b7e0~64bit (-742-8f1b7e0 64bit)

Steps post installation

optional arguments you can pass to freeswitch:
-nf — no forking
-reincarnate — restart the switch on an uncontrolled exit
-reincarnate-reexec — run execv on a restart (helpful for upgrades)
-u [user] — specify user to switch to
-g [group] — specify group to switch to
-core — dump cores
-help — this message
-version — print the version and exit
-rp — enable high(realtime) priority settings
-lp — enable low priority settings
-np — enable normal priority settings
-vg — run under valgrind
-nosql — disable internal sql scoreboard
-heavy-timer — Heavy Timer, possibly more accurate but at a cost
-nonat — disable auto nat detection
-nonatmap — disable auto nat port mapping
-nocal — disable clock calibration
-nort — disable clock clock_realtime
-stop — stop freeswitch
-nc — do not output to a console and background
-ncwait — do not output to a console and background but wait until the system is ready before exiting (implies -nc)
-c — output to a console and stay in the foreground

Options to control locations of files:
-base [basedir] — alternate prefix directory
-cfgname [filename] — alternate filename for FreeSWITCH main configuration file
-conf [confdir] — alternate directory for FreeSWITCH configuration files
-log [logdir] — alternate directory for logfiles
-run [rundir] — alternate directory for runtime files
-db [dbdir] — alternate directory for the internal database
-mod [moddir] — alternate directory for modules
-htdocs [htdocsdir] — alternate directory for htdocs
-scripts [scriptsdir] — alternate directory for scripts
-temp [directory] — alternate directory for temporary files
-grammar [directory] — alternate directory for grammar files
-certs [directory] — alternate directory for certificates
-recordings [directory] — alternate directory for recordings
-storage [directory] — alternate directory for voicemail storage
-cache [directory] — alternate directory for cache files
-sounds [directory] — alternate directory for sound files

Tracing SIP messages and Freeswitch processing for call from external user to internal user .Freeswitch acts as B2BUA

Receives incoming Call INVITE

recv 823 bytes from tcp/[caller_ip]:35365 at 09:55:07.936234:
   ------------------------------------------------------------------------
   INVITE sip:to_number@sometelco.com:5060 SIP/2.0
   Via: SIP/2.0/TCP 192.168.1.23:55934;branch=z9hG4bK-524287-1---cc11593581af6519;rport
   Max-Forwards: 70
   Contact: <sip:from_number@192.168.1.23:55934;transport=tcp>
   To: <sip:to_number@sometelco.com:5060>
   From: "from_number"<sip:from_number@sometelco.com:5060>;tag=47a61272
   Call-ID: 94385YTY3ODNlNzE1YjE5MmY4NmQ3ZWUyZDAzM2E0YzBkM2I
   CSeq: 1 INVITE
   Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO
   Content-Type: application/sdp
   Supported: replaces
   User-Agent: X-Lite release 5.4.0 stamp 94385
   Content-Length: 208

   v=0
   o=- 1553248503383592 1 IN IP4 192.168.1.23
   s=X-Lite release 5.4.0 stamp 94385
   c=IN IP4 192.168.1.23
   t=0 0
   m=audio 49874 RTP/AVP 8 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=sendrecv
   ------------------------------------------------------------------------

checks with ACL for permission and set NAT. Isolate SDP for processing.

New Channel sofia/internal/from_number@sometelco.com:5060 [a8a2003f-5755-40fe-ab63-aab2f5264886]

Running State Change CS_NEW (Cur 1 Tot 274)
receiving invite from caller_ip:35365 version: 1.9.0 -742-8f1b7e0 64bit
IP caller_ip Approved by acl "domains[]". Access Granted.
Setting NAT mode based on nat.auto
Channel sofia/internal/from_number@sometelco.com:5060 entering state [received][100]
Remote SDP:
v=0
o=- 1553248503383592 1 IN IP4 192.168.1.23
s=X-Lite release 5.4.0 stamp 94385
c=IN IP4 192.168.1.23
t=0 0
m=audio 49874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

mainatin and Updates call-state (switch_core_state_machine ) CS_NEW -> CS_INIT -> CS_ROUTING -> RINGING and send 100 trying to caller

State Change CS_NEW -> CS_INIT
State NEW
Running State Change CS_INIT (Cur 1 Tot 274)
State INIT
SOFIA INIT
Standard INIT
State Change CS_INIT -> CS_ROUTING
State INIT going to sleep
Running State Change CS_ROUTING (Cur 1 Tot 274)
Change DOWN -> RINGING
State ROUTING
send 413 bytes to tcp/[caller_ip]:35365 at 09:55:07.937474:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP 192.168.1.23:55934;branch=z9hG4bK-524287-1---cc11593581af6519;rport=35365;received=caller_ip
   From: "from_number"<sip:from_number@sometelco.com:5060>;tag=47a61272
   To: <sip:to_number@sometelco.com:5060>
   Call-ID: 94385YTY3ODNlNzE1YjE5MmY4NmQ3ZWUyZDAzM2E0YzBkM2I
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.9.0-742-8f1b7e0~64bit
   Content-Length: 0

   ------------------------------------------------------------------------

checks dialplan to route incoming call. In this case action is to bridge the incoming call to internal user

mod_sofia.c:154 sofia/internal/from_number@sometelco.com:5060 SOFIA ROUTING
switch_core_state_machine.c:236 sofia/internal/from_number@sometelco.com:5060 Standard ROUTING

mod_dialplan_xml.c:637 Processing from_number <from_number>->to_number in context public
Dialplan: sofia/internal/from_number@sometelco.com:5060 parsing [public->dialplan_cutsom] continue=false
Dialplan: sofia/internal/from_number@sometelco.com:5060 Regex (PASS) [dialplan_cutsom] destination_number(to_number) =~ /^(\d+)$/ break=on-false
Dialplan: sofia/internal/from_number@sometelco.com:5060 Action log(INFO ***** Forwarding calls to gateway ****** ) 
Dialplan: sofia/internal/from_number@sometelco.com:5060 Action bridge({sip_auth_username=user,sip_auth_password=pass,sip_route_uri=sip:to_number@ip_addr;transport=tls,sip_invite_req_uri=sip:to_number@sometelco.com;transport=tls}sofia/external/to_number@ip_addr) 

update call state CS_ROUTING -> CS_EXECUTE

State Change CS_ROUTING -> CS_EXECUTE
State ROUTING going to sleep
Running State Change CS_EXECUTE (Cur 1 Tot 274)
State EXECUTE
SOFIA EXECUTE

set the crypto and codecs for the new call

switch_ivr_originate.c:2159 Parsing global variables
switch_channel.c:1104 New Channel sofia/external/to_number@ip_addr [cc1ae238-9efd-4f51-93e9-05abd48bea4d]
mod_sofia.c:5026 (sofia/external/to_number@ip_addr) State Change CS_NEW -> CS_INIT
switch_core_state_machine.c:584 (sofia/external/to_number@ip_addr) Running State Change CS_INIT (Cur 2 Tot 275)
switch_core_state_machine.c:627 (sofia/external/to_number@ip_addr) State INIT
mod_sofia.c:93 sofia/external/to_number@ip_addr SOFIA INIT
Set Local audio crypto Key [1 AEAD_AES_256_GCM_8 inline:ZbEHd76sP6FZSO9AYcqryybaA4HY3O5p2Uo+e1gmmfVaZCEic6cvKyArhMU]
Set Local video crypto Key [1 AEAD_AES_256_GCM_8 inline:Ehr3LoDR8Ur+wtNAMqoqIDn3S7V2inE2/n++awxS6/1P2ijcqfk12+LM/Pc]
Set Local text crypto Key [1 AEAD_AES_256_GCM_8 inline:NVSfjOmSS5BaP/5yqg+SOXcqvEFTHHrC8R5AYkkClXLuNOXYoaUYlrIWeW0]
Set Local audio crypto Key [2 AEAD_AES_128_GCM_8 inline:ePH/F2Qw5+zi8c7tkBb6Y2AQE5uevp+jWUkjgQ]
Set Local video crypto Key [2 AEAD_AES_128_GCM_8 inline:YWdfNLSx6MqG9WQ3TmsV/cSBDqjRUAbHE0rRCg]
Set Local text crypto Key [2 AEAD_AES_128_GCM_8 inline:DFXOP2V2Ep6FoHNz5HIMrm0cu6Za8I5wOI/hUw]
Set Local audio crypto Key [3 AES_CM_256_HMAC_SHA1_80 inline:SG5rYx3GSR2imutYQ+LzqHufG9UkG3n/SfmFHFOG/r75v2pwf2lG7Qpup+J0mw]
Set Local video crypto Key [3 AES_CM_256_HMAC_SHA1_80 inline:LkU3i9MD25k2wtTfSXUvhlxo66GtMWnXkKoxSdgRZyANoeOhufYnXzbXDo+7+w]
Set Local text crypto Key [3 AES_CM_256_HMAC_SHA1_80 inline:AUgUOVmFunzotvwZ6KuMDnBRR2XKk1DsX2qg465MsT6OAxHc2qKBFpeQEpxrqA]
Set Local audio crypto Key [4 AES_CM_192_HMAC_SHA1_80 inline:2PVBBJEp4QcTzTf4Th8Ag/7KiVPmrYb/FCowiRb6yAuTO/kxQLc]
Set Local video crypto Key [4 AES_CM_192_HMAC_SHA1_80 inline:OiFbZQ6mWuf5sHJT1pFPU6EWxEvQAO/0rcp8uGMf79k7RSR3IQA]
Set Local text crypto Key [4 AES_CM_192_HMAC_SHA1_80 inline:XyednWJmzRfsWQOgdhKaMeOeE/OLmnwo6hVEZWl4OJdKdgK6TVc]
Set Local audio crypto Key [5 AES_CM_128_HMAC_SHA1_80 inline:Yd4L5Qi7A/8xay5ZHWR1jKk9j5Kvy9s2Zo3NOES2]
Set Local video crypto Key [5 AES_CM_128_HMAC_SHA1_80 inline:ImgbbD6cnhnH19O1knP5SSIUULsZTaNJJIUepxt0]
Set Local text crypto Key [5 AES_CM_128_HMAC_SHA1_80 inline:V7+IbSZmTdQNjh/upUZ5TFDSlgarhDTVfV+AcUA+]
Set Local audio crypto Key [6 AES_CM_256_HMAC_SHA1_32 inline:JI+s9uFdZ3JfZmRRfwHr0OrpyZdtUXmMC0WRIZow1EuXRB9xKFRBk6KmSWomqQ]
Set Local video crypto Key [6 AES_CM_256_HMAC_SHA1_32 inline:MX6CGCrMEioUCJsIOCxRqlHOx4mUYRw4DslpY25njZQAkH6MgG/9hp7G8xr44A]
Set Local text crypto Key [6 AES_CM_256_HMAC_SHA1_32 inline:ikCz2sYLGoMO+dlrZj+znlQ3djAkGSYzSLLu6Az8u2THWPgnkFJXVgXSxHOaHw]
Set Local audio crypto Key [7 AES_CM_192_HMAC_SHA1_32 inline:5JzlrMywFZhHuNLWPG/HBrUi/Zcg414Q7ZfSaJQnUF5N9APy+GQ]
Set Local video crypto Key [7 AES_CM_192_HMAC_SHA1_32 inline:K0dZtwH1Q7AuSMBPPUesy047c4nAF+QuFsVvGdf3fYJDOD0Uwxo]
Set Local text crypto Key [7 AES_CM_192_HMAC_SHA1_32 inline:96SwyWAdV1a+BU3UbiX1PHdkRlSS4RtmwPWNPbCR3NDm1MyBh58]
Set Local audio crypto Key [8 AES_CM_128_HMAC_SHA1_32 inline:/RLYPhZs07WCCBRY8tWNTJemT/IFq1VPHGHmGvnG]
Set Local video crypto Key [8 AES_CM_128_HMAC_SHA1_32 inline:mQlgScFq1iMKEW8vobzwhmN9TWSmVblAv9u7c1/c]
Set Local text crypto Key [8 AES_CM_128_HMAC_SHA1_32 inline:WAQveMfrQkPBcfqH2qLmuzY63VLfT+N30/YLyuqE]
Set Local audio crypto Key [9 AES_CM_128_NULL_AUTH inline:f2fx2ekxPG3GTwTYARtquNJ87qO0Q5ei47KYlo9K]
Set Local video crypto Key [9 AES_CM_128_NULL_AUTH inline:qpAkfc1bWnZ0Y/1ql+dNvhIGgxxWZoVltnRD5kqn]
Set Local text crypto Key [9 AES_CM_128_NULL_AUTH inline:LyhSlzI3X38WKPwZ83035Ddvse4J/2KnKoydo2FD]

set proxy route and create SDP for sending invite to bridged client

sofia_glue.c:1268 sip:to_number@ip_addr;transport=tls Setting proxy route to sofia/external/to_number@ip_addr
sofia_glue.c:1299 sofia/external/to_number@ip_addr sending invite version: 1.9.0 -742-8f1b7e0 64bit
Local SDP:
v=0
o=FreeSWITCH 1553228435 1553228436 IN IP4 via_addr
s=FreeSWITCH
c=IN IP4 via_addr
t=0 0
m=audio 20072 RTP/SAVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AEAD_AES_256_GCM_8 inline:ZbEHd76sP6FZSO9AYcqryybaA4HY3O5p2Uo+e1gmmfVaZCEic6cvKyArhMU
a=crypto:2 AEAD_AES_128_GCM_8 inline:ePH/F2Qw5+zi8c7tkBb6Y2AQE5uevp+jWUkjgQ
a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:SG5rYx3GSR2imutYQ+LzqHufG9UkG3n/SfmFHFOG/r75v2pwf2lG7Qpup+J0mw
a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:2PVBBJEp4QcTzTf4Th8Ag/7KiVPmrYb/FCowiRb6yAuTO/kxQLc
a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:Yd4L5Qi7A/8xay5ZHWR1jKk9j5Kvy9s2Zo3NOES2
a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:JI+s9uFdZ3JfZmRRfwHr0OrpyZdtUXmMC0WRIZow1EuXRB9xKFRBk6KmSWomqQ
a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:5JzlrMywFZhHuNLWPG/HBrUi/Zcg414Q7ZfSaJQnUF5N9APy+GQ
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:/RLYPhZs07WCCBRY8tWNTJemT/IFq1VPHGHmGvnG
a=crypto:9 AES_CM_128_NULL_AUTH inline:f2fx2ekxPG3GTwTYARtquNJ87qO0Q5ei47KYlo9K
a=ptime:20
a=sendrecv
m=audio 20072 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

attach the SDP to INVITE and proceed forwarding INVITE to callee

send 1988 bytes to tls/[ip_addr]:5061 at 09:55:07.939831:
   ------------------------------------------------------------------------
   INVITE sip:to_number@sometelco.com;transport=tls SIP/2.0
   Via: SIP/2.0/TLS via_addr:5080;rport;branch=z9hG4bK21Qm9U3eHX0Nc
   Max-Forwards: 69
   From: "from_number" <sip:from_number@via_addr>;tag=8jByBXa2pF1Fj
   To: <sip:to_number@ip_addr>
   Call-ID: 6a827514-c72b-1237-8aab-02a933b32da0
   CSeq: 2070461 INVITE
   Contact: <sip:mod_sofia@via_addr:5080>
   User-Agent: FreeSWITCH-mod_sofia/1.9.0-742-8f1b7e0~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 1162
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "from_number" <sip:from_number@via_addr>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1553228435 1553228436 IN IP4 via_addr
   s=FreeSWITCH
   c=IN IP4 via_addr
   t=0 0
   m=audio 20072 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=crypto:1 AEAD_AES_256_GCM_8 inline:ZbEHd76sP6FZSO9AYcqryybaA4HY3O5p2Uo+e1gmmfVaZCEic6cvKyArhMU
   a=crypto:2 AEAD_AES_128_GCM_8 inline:ePH/F2Qw5+zi8c7tkBb6Y2AQE5uevp+jWUkjgQ
   a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:SG5rYx3GSR2imutYQ+LzqHufG9UkG3n/SfmFHFOG/r75v2pwf2lG7Qpup+J0mw
   a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:2PVBBJEp4QcTzTf4Th8Ag/7KiVPmrYb/FCowiRb6yAuTO/kxQLc
   a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:Yd4L5Qi7A/8xay5ZHWR1jKk9j5Kvy9s2Zo3NOES2
   a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:JI+s9uFdZ3JfZmRRfwHr0OrpyZdtUXmMC0WRIZow1EuXRB9xKFRBk6KmSWomqQ
   a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:5JzlrMywFZhHuNLWPG/HBrUi/Zcg414Q7ZfSaJQnUF5N9APy+GQ
   a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:/RLYPhZs07WCCBRY8tWNTJemT/IFq1VPHGHmGvnG
   a=crypto:9 AES_CM_128_NULL_AUTH inline:f2fx2ekxPG3GTwTYARtquNJ87qO0Q5ei47KYlo9K
   a=ptime:20
   m=audio 20072 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------

manage and update call state for this call leg too CS_INIT -> CS_ROUTING -> CS_CONSUME_MEDIA

Standard INIT
State Change CS_INIT -> CS_ROUTING
State INIT going to sleep
Running State Change CS_ROUTING (Cur 2 Tot 275)
Channel sofia/external/to_number@ip_addr entering state [calling][0]
State ROUTING
SOFIA ROUTING
State Change CS_ROUTING -> CS_CONSUME_MEDIA
State ROUTING going to sleep
Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 275)
State CONSUME_MEDIA
State CONSUME_MEDIA going to sleep
recv 365 bytes from tls/[ip_addr]:5061 at 09:55:07.940977:
   ------------------------------------------------------------------------
   SIP/2.0 100 trying -- your call is important to us
   Via: SIP/2.0/TLS via_addr:5080;rport=59774;branch=z9hG4bK21Qm9U3eHX0Nc;received=via_addr
   From: "from_number" <sip:from_number@via_addr>;tag=8jByBXa2pF1Fj
   To: <sip:to_number@ip_addr>
   Call-ID: 6a827514-c72b-1237-8aab-02a933b32da0
   CSeq: 2070461 INVITE
   Server: XYZ
   Content-Length: 0

   ------------------------------------------------------------------------

Callee from PBX throws auth challenge

recv 483 bytes from tls/[ip_addr]:5061 at 09:55:08.046934:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/TLS via_addr:5080;received=via_addr;rport=59774;branch=z9hG4bK21Qm9U3eHX0Nc
   From: "from_number" <sip:from_number@via_addr>;tag=8jByBXa2pF1Fj
   To: <sip:to_number@ip_addr>;tag=f1cff938000510c1d9006e5a2a4e240b-5736
   Call-ID: 6a827514-c72b-1237-8aab-02a933b32da0
   CSeq: 2070461 INVITE
   Proxy-Authenticate: Digest realm="domain.com", nonce="XJSyI1yUsPf0w1bAocvH4IOCayfWt3bX", qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
send 387 bytes to tls/[ip_addr]:5061 at 09:55:08.047056:
   ------------------------------------------------------------------------
   ACK sip:to_number@sometelco.com;transport=tls SIP/2.0
   Via: SIP/2.0/TLS via_addr:5080;rport;branch=z9hG4bK21Qm9U3eHX0Nc
   Max-Forwards: 69
   From: "from_number" <sip:from_number@via_addr>;tag=8jByBXa2pF1Fj
   To: <sip:to_number@ip_addr>;tag=f1cff938000510c1d9006e5a2a4e240b-5736
   Call-ID: 6a827514-c72b-1237-8aab-02a933b32da0
   CSeq: 2070461 ACK
   Content-Length: 0

   ------------------------------------------------------------------------

calleer sends re-invit with auth details

Authenticating 'altanai' with 'Digest:"doamin.com":altanai:pass'.
send 2273 bytes to tls/[ip_addr]:5061 at 09:55:08.047387:
   ------------------------------------------------------------------------
   INVITE sip:to_number@sometelco.com;transport=tls SIP/2.0
   Via: SIP/2.0/TLS via_addr:5080;rport;branch=z9hG4bK3aHDBQmje6p8Q
   Max-Forwards: 69
   From: "from_number" <sip:from_number@via_addr>;tag=8jByBXa2pF1Fj
   To: <sip:to_number@ip_addr>
   Call-ID: 6a827514-c72b-1237-8aab-02a933b32da0
   CSeq: 2070462 INVITE
   Contact: <sip:mod_sofia@via_addr:5080>
   User-Agent: FreeSWITCH-mod_sofia/1.9.0-742-8f1b7e0~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Proxy-Authorization: Digest username="altanai", realm="domain.com", nonce="XJSyI1yUsPf0w1bAocvH4IOCayfWt3bX", cnonce="apLWcMcrEjerigKpM7MtoA", algorithm=MD5, uri="sip:to_number@sometelco.com;transport=tls", response="0044b00a4d5026252b32eed619d70f9d", qop=auth, nc=00000001
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 1162
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "from_number" <sip:from_number@via_addr>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1553228435 1553228436 IN IP4 via_addr
   s=FreeSWITCH
   c=IN IP4 via_addr
   t=0 0
   m=audio 20072 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=crypto:1 AEAD_AES_256_GCM_8 inline:ZbEHd76sP6FZSO9AYcqryybaA4HY3O5p2Uo+e1gmmfVaZCEic6cvKyArhMU
   a=crypto:2 AEAD_AES_128_GCM_8 inline:ePH/F2Qw5+zi8c7tkBb6Y2AQE5uevp+jWUkjgQ
   a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:SG5rYx3GSR2imutYQ+LzqHufG9UkG3n/SfmFHFOG/r75v2pwf2lG7Qpup+J0mw
   a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:2PVBBJEp4QcTzTf4Th8Ag/7KiVPmrYb/FCowiRb6yAuTO/kxQLc
   a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:Yd4L5Qi7A/8xay5ZHWR1jKk9j5Kvy9s2Zo3NOES2
   a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:JI+s9uFdZ3JfZmRRfwHr0OrpyZdtUXmMC0WRIZow1EuXRB9xKFRBk6KmSWomqQ
   a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:5JzlrMywFZhHuNLWPG/HBrUi/Zcg414Q7ZfSaJQnUF5N9APy+GQ
   a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:/RLYPhZs07WCCBRY8tWNTJemT/IFq1VPHGHmGvnG
   a=crypto:9 AES_CM_128_NULL_AUTH inline:f2fx2ekxPG3GTwTYARtquNJ87qO0Q5ei47KYlo9K
   a=ptime:20
   m=audio 20072 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2019-03-22 09:55:08.041945 [DEBUG] sofia.c:7291 Channel sofia/external/to_number@ip_addr entering state [calling][0]
recv 365 bytes from tls/[ip_addr]:5061 at 09:55:08.048255:
   ------------------------------------------------------------------------
   SIP/2.0 100 trying -- your call is important to us
   Via: SIP/2.0/TLS via_addr:5080;rport=59774;branch=z9hG4bK3aHDBQmje6p8Q;received=via_addr
   From: "from_number" <sip:from_number@via_addr>;tag=8jByBXa2pF1Fj
   To: <sip:to_number@ip_addr>
   Call-ID: 6a827514-c72b-1237-8aab-02a933b32da0
   CSeq: 2070462 INVITE
   Server: XYZ
   Content-Length: 0

   ------------------------------------------------------------------------

Call is accepted by callee , 200 OK is received by Freeswitch PBX

recv 1451 bytes from tls/[ip_addr]:5061 at 09:55:14.223460:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS via_addr:5080;received=via_addr;rport=59774;branch=z9hG4bK3aHDBQmje6p8Q
   Record-Route: <sip:3.92.18.95:5060;lr;ftag=8jByBXa2pF1Fj>
   Record-Route: <sip:18.205.1.67;lr;ftag=8jByBXa2pF1Fj;did=fd.0971>
   Record-Route: <sip:ip_addr:5060;r2=on;lr;ftag=8jByBXa2pF1Fj;nat=yes>
   Record-Route: <sip:ip_addr:5061;transport=tls;r2=on;lr;ftag=8jByBXa2pF1Fj;nat=yes>
   From: "from_number" <sip:from_number@via_addr>;tag=8jByBXa2pF1Fj
   To: <sip:to_number@ip_addr>;tag=D0r5K6pp80Ujm
   Call-ID: 6a827514-c72b-1237-8aab-02a933b32da0
   CSeq: 2070462 INVITE
   Contact: <sip:to_number@34.201.27.78:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.9.0-742-8f1b7e0~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 380
   Remote-Party-ID: "to_number" <sip:to_number@34.201.27.78>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1553215954 1553215955 IN IP4 18.212.123.47
   s=FreeSWITCH
   c=IN IP4 18.212.123.47
   t=0 0
   m=audio 33516 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=sendrecv
   a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:/itE1k5BLMoTNzo7YEv6hCyM6R6wyHem3Coc5jjYVlKR2L3tEzBG5zx1QHgVSg==
   a=ptime:20
   m=audio 0 RTP/SAVP 19
   a=rtpmap:19 
   ------------------------------------------------------------------------

send ACK to callee

Update Callee ID to "to_number" <to_number>
Channel sofia/external/to_number@ip_addr entering state [completing][200]
sofia.c:7301 Remote SDP:
v=0
o=FreeSWITCH 1553215954 1553215955 IN IP4 18.212.123.47
s=FreeSWITCH
c=IN IP4 18.212.123.47
t=0 0
m=audio 33516 RTP/SAVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:/itE1k5BLMoTNzo7YEv6hCyM6R6wyHem3Coc5jjYVlKR2L3tEzBG5zx1QHgVSg==
a=ptime:20
m=audio 0 RTP/SAVP 19

send 953 bytes to tls/[ip_addr]:5061 at 09:55:14.224320:
   ------------------------------------------------------------------------
   ACK sip:to_number@34.201.27.78:5080;transport=udp SIP/2.0
   Via: SIP/2.0/TLS via_addr:5080;rport;branch=z9hG4bK4Ka6cj5NBFDUK
   Route: <sip:ip_addr:5061;transport=tls;r2=on;lr;ftag=8jByBXa2pF1Fj;nat=yes>
   Route: <sip:ip_addr:5060;r2=on;lr;ftag=8jByBXa2pF1Fj;nat=yes>
   Route: <sip:18.205.1.67;lr;ftag=8jByBXa2pF1Fj;did=fd.0971>
   Route: <sip:3.92.18.95:5060;lr;ftag=8jByBXa2pF1Fj>
   Max-Forwards: 70
   From: "from_number" <sip:from_number@via_addr>;tag=8jByBXa2pF1Fj
   To: <sip:to_number@ip_addr>;tag=D0r5K6pp80Ujm
   Call-ID: 6a827514-c72b-1237-8aab-02a933b32da0
   CSeq: 2070462 ACK
   Contact: <sip:mod_sofia@via_addr:5080>
   Proxy-Authorization: Digest username="altanai", realm="domain.com", nonce="XJSyI1yUsPf0w1bAocvH4IOCayfWt3bX", cnonce="apLWcMcrEjerigKpM7MtoA", algorithm=MD5, uri="sip:to_number@sometelco.com;transport=tls", response="0044b00a4d5026252b32eed619d70f9d", qop=auth, nc=00000001
   Content-Length: 0

   ------------------------------------------------------------------------

set audio codecs, update call state CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA

entering state [ready][200]
looking for crypto suite [AEAD_AES_256_GCM_8] in [3 AES_CM_256_HMAC_SHA1_80 inline:/itE1k5BLMoTNzo7YEv6hCyM6R6wyHem3Coc5jjYVlKR2L3tEzBG5zx1QHgVSg==]
looking for crypto suite [AEAD_AES_128_GCM_8] in [3 AES_CM_256_HMAC_SHA1_80 inline:/itE1k5BLMoTNzo7YEv6hCyM6R6wyHem3Coc5jjYVlKR2L3tEzBG5zx1QHgVSg==]
looking for crypto suite [AES_CM_256_HMAC_SHA1_80] in [3 AES_CM_256_HMAC_SHA1_80 inline:/itE1k5BLMoTNzo7YEv6hCyM6R6wyHem3Coc5jjYVlKR2L3tEzBG5zx1QHgVSg==]
Found suite AES_CM_256_HMAC_SHA1_80
Set Remote Key [3 AES_CM_256_HMAC_SHA1_80 inline:/itE1k5BLMoTNzo7YEv6hCyM6R6wyHem3Coc5jjYVlKR2L3tEzBG5zx1QHgVSg==]
Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
Set telephone-event payload to 101@8000
Set Codec sofia/external/to_number@ip_addr PCMA/8000 20 ms 160 samples 64000 bits 1 channels
sofia/external/to_number@ip_addr Original read codec set to PCMA:8
Set telephone-event payload to 101@8000
sofia/external/to_number@ip_addr Set 2833 dtmf send payload to 101 recv payload to 101
AUDIO RTP [sofia/external/to_number@ip_addr] 10.130.74.15 port 20072 -> 18.212.123.47 port 33516 codec: 8 ms: 20
Starting timer [soft] 160 bytes per 20ms
Set 2833 dtmf send payload to 101
Set 2833 dtmf receive payload to 101
Set rtp dtmf delay to 40
Activating audio Secure RTP SEND
srtp:sdes:AES_CM_256_HMAC_SHA1_80
Activating audio Secure RTP RECV
srtp:sdes:AES_CM_256_HMAC_SHA1_80
has been answered
Callstate Change DOWN -> ACTIVE
Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
Set telephone-event payload to 101@8000
Set Codec sofia/internal/from_number@sometelco.com:5060 PCMA/8000 20 ms 160 samples 64000 bits 1 channels
sofia/internal/from_number@sometelco.com:5060 Original read codec set to PCMA:8
Set telephone-event payload to 101@8000
sofia/internal/from_number@sometelco.com:5060 Set 2833 dtmf send payload to 101 recv payload to 101
AUDIO RTP [sofia/internal/from_number@sometelco.com:5060] 10.130.74.15 port 29426 -> 192.168.1.23 port 49874 codec: 8 ms: 20
Starting timer [soft] 160 bytes per 20ms
Set 2833 dtmf send payload to 101
Set 2833 dtmf receive payload to 101
Set rtp dtmf delay to 40

send early media/ RTP to Callee

 Pre-Answer sofia/internal/from_number@sometelco.com:5060!
 Callstate Change RINGING -> EARLY
 2019-03-22 09:55:14.221933 [DEBUG] switch_core_media.c:8147 Audio params are unchanged for sofia/internal/from_number@sometelco.com:5060.
 2019-03-22 09:55:14.221933 [DEBUG] mod_sofia.c:881 Local SDP sofia/internal/from_number@sometelco.com:5060:
 v=0
 o=FreeSWITCH 1553219088 1553219089 IN IP4 via_addr
 s=FreeSWITCH
 c=IN IP4 via_addr
 t=0 0
 m=audio 29426 RTP/AVP 8 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv
Pre-Answer sofia/internal/from_number@sometelco.com:5060!
Callstate Change RINGING -> EARLY
2019-03-22 09:55:14.221933 [DEBUG] switch_core_media.c:8147 Audio params are unchanged for sofia/internal/from_number@sometelco.com:5060.
2019-03-22 09:55:14.221933 [DEBUG] mod_sofia.c:881 Local SDP sofia/internal/from_number@sometelco.com:5060:
v=0
o=FreeSWITCH 1553219088 1553219089 IN IP4 via_addr
s=FreeSWITCH
c=IN IP4 via_addr
t=0 0
m=audio 29426 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

send 1254 bytes to tcp/[caller_ip]:35365 at 09:55:14.232934:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TCP 192.168.1.23:55934;branch=z9hG4bK-524287-1---cc11593581af6519;rport=35365;received=caller_ip
   From: "from_number"<sip:from_number@sometelco.com:5060>;tag=47a61272
   To: <sip:to_number@sometelco.com:5060>;tag=NjvKFKQaHp52e
   Call-ID: 94385YTY3ODNlNzE1YjE5MmY4NmQ3ZWUyZDAzM2E0YzBkM2I
   CSeq: 1 INVITE
   Contact: <sip:to_number@via_addr:5060;transport=tcp>
   User-Agent: FreeSWITCH-mod_sofia/1.9.0-742-8f1b7e0~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Session-Expires: 120;refresher=uas
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 220
   Remote-Party-ID: "to_number" <sip:to_number@sometelco.com>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1553219088 1553219089 IN IP4 via_addr
   s=FreeSWITCH
   c=IN IP4 via_addr
   t=0 0
   m=audio 29426 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
entering state [completed][200]
Channel [sofia/internal/from_number@sometelco.com:5060] has been answered
Callstate Change EARLY -> ACTIVE
Originate Resulted in Success: [sofia/external/to_number@ip_addr]
State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 275)
State EXCHANGE_MEDIA
SOFIA EXCHANGE_MEDIA
recv 507 bytes from tcp/[caller_ip]:35365 at 09:55:14.459247:
   ------------------------------------------------------------------------
   ACK sip:to_number@via_addr:5060;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 192.168.1.23:55934;branch=z9hG4bK-524287-1---104aee5ed0b7ca66;rport
   Max-Forwards: 70
   Contact: <sip:from_number@192.168.1.23:55934;transport=tcp>
   To: <sip:to_number@sometelco.com:5060>;tag=NjvKFKQaHp52e
   From: "from_number"<sip:from_number@sometelco.com:5060>;tag=47a61272
   Call-ID: 94385YTY3ODNlNzE1YjE5MmY4NmQ3ZWUyZDAzM2E0YzBkM2I
   CSeq: 1 ACK
   User-Agent: X-Lite release 5.4.0 stamp 94385
   Content-Length: 0

   ------------------------------------------------------------------------