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Asterisk Complete Training

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Flavio Eduardo de Andrade Goncalves, profile picture
Flavio Eduardo de Andrade Goncalves

This document provides an overview and objectives for an E-Learning training course on building a complete PBX with Asterisk. The training will cover understanding and installing Asterisk, building a simple PBX with SIP phones and SIP trunks, configuring features like dialplans, IVRs and voicemail, and integrating applications. By the end of the course, students will have hands-on experience building their own basic Asterisk system. Various chapters include objectives, presentations on topics, and instructions for labs to gain practical experience configuring Asterisk.

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E-Learning Complete Deck of Slides of the Complete Asterisk Training You can enroll at: http://bit.ly/2E6U7fP
E-Learning Books, eBooks and Other Trainings From the same Author Online Trainings Complete Asterisk Training Coupon http://bit.ly/2E6U7fP Understanding and Troubleshooting SIP http://bit.ly/2WZKkzy VoIP Design Training Coupon http://bit.ly/2tlLFmG VoIP Hacking Training Coupon http://bit.ly/2GL5MCT Quick Start to OpenSIPS Coupon http://bit.ly/2Bt72XJ Books Complete Asterisk Training Paperback and Kindle Book https://amzn.to/2tm7TFb Complete Asterisk Training eBook PDF http://bit.ly/2UUebHG Building Telephony Systems with OpenSIPS http://bit.ly/2SsUt9a
E-Learning Welcome Building a complete PBX with Asterisk
E-Learning • By the end of this training you should be able to: – Understand what is Asterisk and where it can be applied – Choose the appropriate hardware and software for your project – Install Asterisk – Build a simple PBX with SIP phones and SIP trunks – Call between phones to a SIP trunk and from a SIP trunk – Configure an IVR and Auto-Attendant to receive calls – Configure a Conference Room – Configure a VoiceMail Server – Understand how to integrate applications with Asterisk Objectives of the training
E-Learning • Most chapters have Labs • You will need Virtual Box installed in your notebook or laptop • At the end of the training you should have a simple system ready to run Practical Training
E-Learning Introduction to Asterisk
E-Learning • By the end of this chapter you should be able to: – Describe what Asterisk is – Understand the Asterisk ecosystem – Select a hardware/software for your project – Understand the Asterisk arquitecture – Point common usage scenarios – Find documentation and information Objectives
E-Learning What is it? Asterisk is a PBX implemented as an open source software. It was originally created by Mark Spencer in 1999. As any other PBX it allows you to connect phones and make calls. With the passage of time Asterisk has becoma a major telephony platform for applications such as Dialers, Call Centers, Interactive Voice, Response, SoftSwitches. The imagination is the limit.
E-Learning
E-Learning
E-Learning
E-Learning Licensing https://www.asterisk.org/products/software
E-Learning Which hardware to use? Phone Company IP Phones Analog Telephone Adapters ISDN and Analog Lines SIP Provider Internet Telephony Service Provider Softphones
E-Learning Which software to use?
E-Learning Asterisk Versions https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
E-Learning Asterisk Architecture
E-Learning Asterisk Applications API Asterisk Channels API FileformatAPI CodecTranslationAPIs chan_dahdi (Analog and Digital lines), chan_sip (SIP channels), chan_iax (IAX channels) Gsm,ALaw,Ulaw,G.723,G729, ADPCM,MP3,Speex,LPC10 GSM,WAV,G723af,MP3 VoiceMail, Conference, Dial, Background Codec Translator Switching Core Application Launcher IO scheduler and Manager Dynamyc Module Loader
E-Learning Channels
E-Learning Codecs G729 SLIN SLING729 SLIN iLBC SLIN iLBC Codec Bandwidth Comment G.711 ulaw 64 Kbps Excelent G.711 alaw 64 Kbps Excelent G.723.1 – Modo 5.3/6.3 Kbps Pass-thru only G.726 32 Kbps Excelent G.729 8 Kbps Excelent (Requires license) GSM 13 Kbps Good Quality iLBC 15-18 Kbps Good Quality G.722 64 Kbps High Definition Opus 5.3-512 Kbps Narrowband to wideband
E-Learning Protocols • SIP; • H323; • IAXv1 e v2; • MGCP; • SCCP (Cisco Skinny); • Nortel Unistim
E-Learning File formats
E-Learning Applications cli>core show aplications
E-Learning Use Scenarios
E-Learning Lab. One Phone, One Line PBX
E-Learning Enterprise Solutions
E-Learning Connecting headquarters and branches
E-Learning Asterisk as an application server TELCO E1/Analog E1/Analog Asterisk PBX IVR Conference VoiceMail
E-Learning Asterisk as a Contact Center 1-800 Agent1 Agent2 AgentN DAC Ringall,RoundRobin,Rrmemory, LeastRecent,Priority Call Queue
E-Learning Asterisk vs. Conventional system IP Phones Softswitch PSTN Gateway VoiceMail Music On Hold Analog Telephone Adapter CISCO IP PHONE 7905SERIES 1 2 A B C 3 D E F 4 5 J K L 6 M N OGH I 7 8 T U V 9 W X Y ZP Q R S * 0 # 4 7 P QRS * CISCO IPPHONE 7905SERIE S 1 2 A B C 3 DE F 4 5 J KL 6 M N OG HI 7 8 TU V 9 W X YZP QR S * 0 # 4 7 PQRS * CISCO IPPHONE 7905SERIE S 1 2 A B C 3 DE F 4 5 J KL 6 M N OG HI 7 8 TU V 9 W X YZP QR S * 0 # 4 7 PQRS * Telco or PBX Telco or PBX CISCO IP PHONE 7905SERIES 1 2 A B C 3 DE F 4 5 J K L 6 M N OG H I 7 8 T U V 9 W XY ZP QR S * 0 # 4 7 P QRS * IP Phones Softswitch Gateway for Public Networks Unified Messaging System Gateway for Analog Telephony CISCO IP PHONE 7905SERIES 1 2 A B C 3 DE F 4 5 J K L 6 M N OG H I 7 8 T U V 9 W XY ZP QR S * 0 # 4 7 P QRS * CISCO IP PHONE 7905SERIES 1 2 A B C 3 DE F 4 5 J K L 6 M N OG H I 7 8 T U V 9 W XY ZP QR S * 0 # 4 7 P QRS *
E-Learning Where to find documentation • wiki.asterisk.org • www.asterisk.org • www.voip-info.org
E-Learning • AGI – Asterisk Gateway Interface • AMI – Asterisk Managment Interface • ARI – Asterisk Rest Interface Integration with other systems
E-Learning Summary Arquitecture Who? Where? When? Why? Usage scenarios
E-Learning Downloading and Installing Asterisk
E-Learning • By the end of this chapter you should be able to: – Dimension the required hardware – Install – Start and Stop – Connect to the console – Start Asterisk in the Linux boot Objectives
E-Learning • Labs, 1GB RAM, 10GB disk • Virtual Box, Bridge Mode • Ubuntu 16.04, 32 bits Minimum hardware
E-Learning • Empiric Process • Use SIPP to test exceptions • General Rules (80/20) – 80% or more of the Asterisk users have less than 30 simultaneous calls and no more than 240 users. Any modern server could take this load. – If you belong to the other 20%, test using SIPP. Use a large security coefficient. So if the test got 3000 simultaneous calls, use a SC=4 and dimension for no more than 750. Dimensioning
E-Learning • Simultaneous calls • Registered devices • Transcoding – Pay attention to high complexity codecs 1G729 = 13 MIPS, 1 narrowband OPUS=30 Mflops. • Call Attempts per second (CAPS) – Load with more tha 100 CAPS can be problematic Impairment factors
E-Learning • CDRs in a database – Each CDR generates I/Os for data and indexes – SATA HD supports only 75 IOPS • Recording. Recording is intensive in terms of disk I/O – Try to keep below 200 in a SAS 15K RPM disk • Applications – Conference Rooms, Queues and IVRs affect the performance in an unpredictable way. It is not possible to predict the performance of a unknow application before it has been developed Hardware notes
E-Learning • Sipp – Traffic Generatos • Sippy Cup - http://mojolingo.github.io/sippy_cup/ Testing tools
E-Learning • From te authors experience – Use a good network switch and good interface cards. – Use Echo Cancelling by hardware (cards) – Use a Transcoding card if you plan to transcode a large amount of calls Tips
E-Learning Astricon 2017 • Tests on a low cost mini J1900 4GBRAM
E-Learning Echo test with media • Command line • sipp -s 1000 10.8.1.46:5060 -sf sipp_pcap.xml -i 10.8.1.46 -d 90000
E-Learning Echo Test with media calls per second 37 35 33 32 32 29 30 31 32 33 34 35 36 37 38 Asterisk 11 Asterisk 13 Asterisk 15 chan_sip chan_pjsip
E-Learning Echo Test with media simultaneous calls 3338 3076 2820 2700 2700 0 500 1000 1500 2000 2500 3000 3500 4000 Asterisk 11 Asterisk 13 Asterisk 15 chan_sip chan_pjsip
E-Learning • Asterisk requires the following packages – Asterisk • Optionals – dahdi-linux – drivers das placas – dahdi-tools – utilitários para as placas – libpri – usado para placas ISDN Installation Packages
E-Learning • From the source code – Flexible – Optmized • From packages – No official maintainers – Usually not the latest version – Simple and quick • Distro – FreePBX – AsteriskNOW How to install?
E-Learning • The variety is huge • Trend – Use SIP trunking , forget cards • If you don't have access to a SIP trunk – Digium, Sangoma, OpenVOX, Aligera... – Alternative cheap hardware (only If you have a hacker soul ) • CHAN_DONGLE https://github.com/bg111/asterisk-chan-dongle • X100P (1 FXO) Telephony Cards
E-Learning Demo
E-Learning • In this chapter you've learned how to – Dimansion the hardware required for Asterisk – Install Asterisk – Start and Stop Linux – Start Asterisk at the boot – Connect to the console Summary
E-Learning Building an IP PBX Section Overview
E-Learning • Part I – SIP extensions • Part II – SIP trunks • Part III – Dialplan • Part IV – Adding features Sections
E-Learning By the end of this section you should be able to: • Build a PBX with the following characteristics – Support IP phones based on SIP – Connect to a VoIP operator using a SIP trunk – Dial between extensions – Dial to external destination using 9 as a prefix – Missed calls sent to the voicemail – Conference room in the extension 5 – Incoming calls directed to an Operator or AutoAttendant Objectives
E-Learning Grammar Object Creation Conf. File Example Simple group All in the same line extensions.conf exten=>4000,1,Dial(SIP/4000) Inheritance Options defined before object declaration chan_dahdi.conf [channels] context=default signalling=fxs_ks group=1 channel => 1 Complex entities Each entity receives a context sip.conf, iax.conf [cisco] type=friend secret=mysecret host=10.1.30.50 context=from-internal [xlite] type=friend secret=xlite host=dynamic Context=from-internal Asterisk Grammar
E-Learning LAB
E-Learning Installation Sequence Overview © 2009 Virtual Office Com. E Ind. Ltda. Todos os direitos reservados. Estes slides fazem para do curso Asterisk Básico www.voffice.com.br/elearning
E-Learning Building the PBX part I Extensions
E-Learning IP Phone Configuration Analog Telephony Adapters Softphones ÌP Phones
E-Learning sip.conf, General Section [general] bindport = 5060 bindaddr = 192.168.0.1 context = dummy allowguest=no alwaysauthreject=yes disallow = all allow = ulaw seção geral do arquivo sip.conf
E-Learning sip.conf, Devices section [zoiper] type=friend secret=#m4rksp3nc3r# host=dynamic nat=yes qualify=yes directmedia=no context=from-internal deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/255.255.0.0 [xlite] …
E-Learning LAB 3-1 Configuring the phones
E-Learning Building the PBX part II Trunks
E-Learning • Becoming mainstream on Telcos • Digital, high volume only limited to bandwidth • No investment on legacy hardware • Savings on rack space and eletricity SIP trunks
E-Learning • Providers's IP address • Name and Password • Some allow IP authentication • Authentication per transaction – Register – Calls • Return Address – Which extension to ring when receiving an incoming call SIP trunk parameters
E-Learning Asterisk Registration register => login:passwd@domain:port/9999 Name Password IP or domain UDP port Callback Extension
E-Learning • sip show registry Checking the registration?
E-Learning • RFC2833 – DTMF as named events • INFO – SIP Info message • Audio – DTMF as audio in the channel DTMF
E-Learning [general] srvlookup=yes register => login:pass@domain:port/9999 [siptrunk] type=peer defaultuser=login secret=senha port=5060 insecure=invite host=dominio fromuser=login fromdomain=dominio dtmfmode=rfc2833 context=from-sip disallow=all allow=ulaw Creating a SIP trunk
E-Learning Lab 3-2 Creating a SIP trunk
E-Learning Building the PBX part III Dialplan
E-Learning • File /etc/asterisk/extensions.conf • Extensions format • Contexts, extensions and priorities • Call flow between contexts • Dialing between extensions • Dialing external numbers • Incoming calls • Voicemail and conference rooms Building a dialplan
E-Learning Dialplan configuration /etc/asterisk/extensions.conf
E-Learning
E-Learning Call processing [globlals] OPERATOR=SIP/4000 [incoming] exten=s,1,dial(${OPERATOR},20) exten=s,n,hangup() [default] ;dialing other extensions starting with 4 followed by three digits exten=>_4XXX,1,Dial(SIP/${EXTEN},20) exten=>_4XXX,n,hangup() ; Send every digit after 9 to the PSTN exten=>_9.,1,Dial(ZAP/g1/${EXTEN:1},20) exten=>_9.,n,hangup() [4001] type=friend context=default extensions.confchan_dahdi.conf sip.conf
E-Learning exten=>number(name),{priority|label{+|-}offset}[(alias)],aplication() 8000 Numeric Alexander Alphanumeric 4321/ 1234 Numeric with CallerID _4XXX Regular Expression s Standard extensions Extensions Format Priorities Format 1-9 Priorit number n next s same n+/-x n+x, n-x s+/-x s+x, s-x hint Presence related Extensions Syntax
E-Learning Simple example exten=>123/100,1,Answer() exten=>123/100,2,Playback(tt-weasels) exten=>123/100,3,Hangup()
E-Learning Pattern Examples _ (Underscore) Pattern start character . (point) Wildcard one or more digits ! (exclamation) Wildcard one or more digits immediatey (ISDN) [123-7] Any values in brackets (1,2,3a7) X Any digit from 0-9 Z Any digit from 1-9 N Any digit from 2-9
E-Learning Asterisk Standard Extensions i : Invalid s : Start h : Hangup t : Timeout T : AbsoluteTimeout o : Operator a : used when “*” is pressed in the voicemail menu fax : used for fax detection Talk : used for backgroundDetect() Standard extensions
E-Learning • Global set(Global(variable)=content) • Channel ${EXTEN} Called extension ${CONTEXT} Current context ${CALLERID} Current callerID ${PRIORITY} Current Priority • Application ${DIALEDTIME} ${ANSWEREDTIME} ${DIALSTATUS} CHANUNAVAIL CONGESTION NOANSWER BUSY Variables
E-Learning Math operators Add (+), Subtract (-), Multiply (*), Division (/), Modulus (%) Logica operator Logical “AND” (&), logical “OR” (|), denial (!) Comparison operators (=, >, >=, <,<=,!=) Regular expression operators Regular expression matching (:), Regular expression exact matching (=~) Conditional operators expression1 ? expression2 :: expression3 Expressions Expressions
E-Learning Operators, Functions and Substrings Example: exten=>100,1,Set(Fruit=Orange) exten=>100,2,NoOp(${LEN(Fruit)}) exten=>100,3,NoOp(${LEN(${Fruit})}) Eamples: ${string:offset:length} ${123456789:1}-returns 23456789 ${123456789:-4}-returns 6789 ${123456789:0:3}-returns 123 ${123456789:2:3}-returns 345 ${123456789:-4:3}-returns 678 Example; $[${I}+100]
E-Learning Applications Applications to build a simple dial plan  answer() – Answers a channel  dial() – Dial another channel  hangup() – Hangup a channel  playback() – Play na audio file  goto() – Jump to the next priority, extension or context
E-Learning The application Dial() ;Dialing a channel Dial(type/identifier,timeout,options, URL) ;Dialing to multiple channels Dial(Technology/resource[&Tech2/resource2...][|timeout][|opt ions][|URL]): exten=>_6XXX,1,Dial(SIP/${EXTEN},60,Tt)
E-Learning Building a PBX, contexts Context Description from-siptrunk Calls coming from the siptrunk from-internal All extensions SIP and IAX
E-Learning Building the PBX Dialing between extensions [from-internal] exten=>6000,1,dial(SIP/pap21A,20) exten=>6001,1,dial(SIP/xlite1,20) Dialing to the PSTN prefixing 9 [from-internal] ;Dialing to the PSTN via SIP trunk exten=>_9.,1,dial(SIP/siptrunk/${EXTEN:1},20) CLI>dialplan reload
E-Learning Incoming Call from the SIP trunk [globals] OPERATOR=>SIP/6000 [from-siptrunk] exten=>9999,1,dial(${OPERATOR},20) Why 9999? Remember: register => login:pass@domain:port/9999 In the register command you define the extension called for incoming calls
E-Learning Lab 3-3 Building the PBX
E-Learning Building the PBX part IV IVR, Conference and Voicemail
E-Learning • By the end of this chapter you will be able to: – Record prompts directly from your phone – Build an autoattendant – Build a simple IVR – Send missed calls to a voicemail – Send calls to an audioconference room Objectives
E-Learning Recording Messages in a file [from-internal] exten => _4.,1,Record(${EXTEN:1}:gsm) exten => _4.,n,wait(1) exten => _4.,n,Playback(${EXTEN:1}) exten => _4.,n,Hangup() Dialing 4100 will record 100.gsm Dialing 4menu will record menu.gsm Dialing 4greeting will record greeting.gsm
E-Learning Auto-attendant [from-siptrunk] Include=autoattendant [autoattendant] exten=>9999,1,answer() exten=>9999,2,background(greeting) exten=>9999,3,Dial(${OPERATOR}) exten=>6000,1,Dial(SIP/xlite) exten=>6001,1,Dial(SIP/zoiper) Background(filename1[&filename2...][|options[|langoverride][|context]]) s – skip the message n – Does not answer the channels before playing files (183 Early Media) m – Stops when a digit matches the one digit extension in the destination context
E-Learning Using an IVR [from-siptrunk] include=>ivr [ivr] exten=>9999,1,answer() exten=>9999,n,background(greeting) exten=>9999,n,Dial(${OPERATOR}) exten=>1,1,dial(SIP/zoiper) exten=>2,1,dial(SIP/xlite) exten=>3,1,dial(SIP/blink)
E-Learning Voicemail and Conference Voicemail general configuration (not necessary for users.conf) [general] [default] <Mailbox ID> = <pincode>, <fullname>, <email address>, <pagermail>, <option=value>, <option=value> 1234=>1234, Joe Doe, joedoe@somecompany.com, joedoe@somecompany.com, saycid=yes [stdexten] exten=>s,1,Dial(${ARG1},20,tT) exten=>s,n,Goto(${DIALSTATUS}) exten=>s,n,hangup() exten=>s,n(BUSY),voicemail(${ARG2},b) exten=>s,n,hangup() exten=>s,n(NOANSWER),voicemail(${ARG2},u) exten=>s,n,hangup() exten=>s,n(CANCEL),hangup exten=>s,n(CHANUNAVAIL),hangup exten=>s,n(CONGESTION),hangup [from-internal] exten=>6000,1,Gosub(stdexten,s,1(SIP/6000,${EXTEN})) exten=>6001,1,Gosub(stdexten,s,1(SIP/6001,${EXTEN}))
E-Learning Voicemailmain() exten=>9000,1,VoiceMailMain() Level#1 0 Mailbox options 1 Read Voicemail messages 2 Change folders * Help # Exit 1 Record your unavailable message 2 Record your busy message 3 Record your name 4 Record your temporary message 5 Change your password * return to the main menu 3 Advanced Options 4 Play previous message 5 Repeat current message 6 Play next message 7 Delete current message 8 Forward message to another mailbox 9 Save message in folder 0 inbox 1 old 2 work 3 family 4 friends 5 cust1 6 cust2 1 Accept 2 Review 3 Re-record 0 Reach operator 1 Reply 2 Call Back 3 Envelope 4 Outgoing call VoiceMailMain() Menus
E-Learning Conference ConfBridge Conference bridge application. [Description] Enters the user into a specified conference bridge. The user can exit the conference by hangup or DTMF menu option. This application sets the following channel variable upon completion: ${CONFBRIDGE_RESULT}: FAILED:The channel encountered an error and could not enter the conference. HANGUP:The channel exited the conference by hanging up. KICKED:The channel was kicked from the conference. ENDMARKED:The channel left the conference as a result of the last marked user leaving. DTMF:The channel pressed a DTMF sequence to exit the conference. TIMEOUT:The channel reached its configured timeout. [Syntax] ConfBridge(conference[,bridge_profile[,user_profile[,menu]]]) exten=5,1,Confbridge(main)
E-Learning Summary Ending this section you are now able to: • Understand and build configuration files • Build a simple PBX • Install and configure a SIP trunk • Build a simple dial plan • Dial between extensions and external destinations • Receive calls in the operator • Build a simple autoattendant system
E-Learning Lab 3-4 Adding features
E-Learning PBX Features Overview
E-Learning Objectives • By the end of this chapter you should be able to use: – Call Transfer – Call Pickup – Three-way conference – Call Parking – Call Recording – Music On Hold – Follow me
E-Learning Asterisk Music on hold Call transfer Call parking Call pickup Call recording Dialplan Follow-me Blacklist In the phone Call hold Call transfer Three-way conference Message waiting indicator Where resources are implemented
E-Learning Call Transfer Blind transfer  You need to enable in the dial() applicationwith the options “tT”;  Press “#” during the call;  Dial the extension to transfer to  Hangup the call  If the call does not go thru, it will ring back Attended Transfer  Enabled on features.conf (disabled by default);  Dial “*2” to start the transfer;  Dial the destination extension  Talk to the person on the destination  Hangup the call  The system transfers the call
E-Learning Call Parking 701 702 703 704 705 706 707 708 709 710 711 712 713 714 715 716 717 718 719 720 700 Parking Lot 1. Transfer the call to the extension 700. 2. The call is parked in the first free slot. The slot is announced to you. 3. Call the extension where the call was parked and you resume the call.
E-Learning Call Pickup callgroup=1 pickupgroup=1 callgroup=2 pickupgroup=2 callgroup=3 pickupgroup=1,2,3 (Operator) Operator capture calls from groups 1,2 and 3 Members capture calls only inside their group Sales P&D
E-Learning Call Recording • Call MixMonitor before the Dial command. • In line recording – automixmon – Don't forget the X option in the Dial() command MixMonitor(<file>.<ext>[|<options>[|<command>]]) Records The audio on the current channel to the specified file. Valid Options: b – Only save audio to the file while the channel is bridged. *does not include conferences* a – Append to the file instead of overwriting it. v(<x>) – Adjust the heard volume by a factor of <x> -4/4. V(<x>) – Adjust the spoken volume by a factor of <x> -4/4. W(<x>) – Adjust the overall volume by a factor of <x> -4/4.
E-Learning • Music on hold is organized in classes Music on Hold [default] -> name of the class mode=files -> files (files,custom,mp3) directory=moh ->directory where to found the recording in Asterisk format Applications MusicOnHold(class,[duration]) -> Play music on hold Set(CHANNEL(musicclass)=…)-> Set the class for other applications such as Dial, Queue...
E-Learning • Application to forward a call FollowMe [6000] context => from-internal number => 9130523456789,30 number => 9130523456788,30 /etc/asterisk/followme.conf [from-internal] exten=6000,1,dial(SIP/Zoiper) exten=6000,n,followme(6000) /etc/asterisk/extensions.conf
E-Learning • In this lecture you have learned how: – To transfer calls using the Transfer Button – To transfer calls using *2 in the PBX – To do attended and unattended transfers – To pick-up a call – To park calls – To create a 3way conference Summary
E-Learning Advanced Topics on SIP
E-Learning • There are some SIP skills that can be useful. • In this section we will present some of the skills – To use PJSIP with Asterisk 15 (chan_sip will be deprecated in the near future) – chan_sip in depth Peer matching Channel naming conventions – NAT traversal Connecting phones behind NAT ALG workarounds Install Asterisk in the cloud behind NAT Section overview
E-Learning • Four chapters borrowed from the training (Understanding and TroubleShooting SIP) – Introduction to SIP – SIP addresses and headers – Media and Codec Selection • chan_pjsip • chan_sip • NAT traversal – Running clients behind NAT – Workarounds for SIP ALG – Running Asterisk in the Cloud behind NAT Section development
E-Learning Using PJSIP
E-Learning • New features – Multiple registrations – Multiple transports – API friendly, easier to develop and extend • Problems with chan_sip – Monolythic – No oficial support, only community – Can be officially deprecated in the next version • Problems with chan_pjsip – low in adoption – Configuration is more complex Why?
E-Learning Advanced Topics in the Dialplan
E-Learning • What is important here? – Context inclusion – Dialplan processing order – Time contexts and GotoIfTime – General section variable autofallthrough – In depth look at applications – Persisting data with AstDB – Limiting simultaneous calls – Applications extensions language Section Overview
E-Learning [local] exten=_9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) [ld] exten=_91N.,Dial(Zap/g1/${EXTEN:1}) include=>local [internat] exten=_9011.,1,Dial(Zap/g1/${EXTEN:1}) include=>ld Context inclusion sip.conf extensions.conf Context inclusion
E-Learning Dial plan processing order 1. An exact match using dialed number and callerID 2. An exact match using only dialed number 3. A pattern that matches the dialed number 4. A context included in the switch statement 5. An included context Dialplan processing order
E-Learning Time based contexts Time based contexts
E-Learning Generating Call Detail Record - CDRs
E-Learning • What is important here? – Why generate CDRs – Hot to generating CDRs – How to add extra fields – How to control, reset and avoid CDRs – How to send CDRS to a database Section Overview
E-Learning • Creation – Channel is created – Leaves a bridge and is not hungup – Forked from a prior record – Enters a multi-party bridge • Finalized – Dial completes with status different than ANSWER – If either party leaves the bridge – if either channel hangs up – if CDR is forked When CDRS are generated
E-Learning CDR Format Field Type Description Access accountcode String (20) An account code r/w src String (80) The Caller ID Number r dst String (80) The destination extension r dcontext String (80) The destination context r clid String (80) The Caller ID with text r channel String (80) The name of the Party A channel r dstchannel String (80) The name of the Party B channel r lastapp String (80) The last application the Party A executed r lastdata String (80) The application data for the last application r start Date/time The time the CDR was created r answer Date/time The time when Party A was answered r end Date/time The time when the CDR was finished. r duration Integer The time in seconds from start until end r billsec Integer The time in seconds from answer until end r disposition Enum The final known disposition r amaflags Enum A flag specified on the Party r/w uniqueid String (32) A unique identifier for the Party A channel r
E-Learning • /var/log/asterisk/Master.csv Standard CDR
E-Learning Troubleshooting Asterisk
E-Learning • Troubleshooting Methodology • How to produce and get data with logs • Where to get help • How to get SIP traces Section Overview
E-Learning • What is important here? – What to do when Asterisk does not start – What to do when a Module does not run – NoOP() and Verbose() – Where to find logs and how to rotate them – How to enable verbose and debug – Hot to get help Troubleshooting Asterisk
E-Learning • Use asterisk –vvvc – Open in console mode, check if some module is not loading What to do when Asterisk does not start?
E-Learning • When a module does not load, its commands don't exist. – no such command sip show peers is an example of this case. – Try to load manually using module load chan_sip.so – Check for UDP/TCP port conflicts using netstat -tanpu What to do when a module does not load
E-Learning Best practices in Asterisk Security
E-Learning Agenda 1. How big is the problem? 2. Anatomy of an attack. 3. Types of attacks 4. How to reduce damage if all previous measures failed
E-Learning How big is the problem? • AT&T case, 1 million dollar phone bill Source: https://www.huffpostbrasil.com/entry/michael-smith-att-phone-bill_n_1659136 • Architecture firm received a phone bill of USD166.000,00 https://www.nytimes.com/2014/10/20/technology/dial-and-redial-phone-hackers-stealing-billions-.html
E-Learning Anatomy of a simple attack. Step 1 – Buy a Premium Rate Number Step 2 – Find a vulnerable VoIP device And call the premium rate number Step 3 – Withdrawn in the premium number
E-Learning
E-Learning Common ways to get a password 1. SIP Scan and Bruteforce 2. TFTP attacks 3. Phone vulnerabilities 4. PBX web interface vulnerabilities 5. Operating System vulnerabilities Shell Shock, Heartbleed
E-Learning Under Heavy Attack! • Basic Scan – sipvicious, friendly-scanner • Distributed SCAM by W32.Sality virus (discovered by Symantec/2010) Source: Symantec http://www.symantec.com/connect/blogs/distributed-cracker-voip Peer-to-Peer Botnet Thousands of Corporate PBXs SIP Scan. Thousands of Register Attempts.
E-Learning SIP Scan Mitigation • Mandatory strong passwords – 8 digits minimum, special chars… • Detect multiple authentication failures – Block IP with Fail2Ban • Early detection and discard – Detect specific signatures and patterns (IPTABLES)
E-Learning TFTP Attack • Trivial Attack against VoIP Infrastructure – 1st Option bruteforce tftp server – 2nd Option sniff tftp files using MitM techniques TFTP Server Get file 0001234A5B6C.cfg Plaintext Configuration File With credentials in plain text XML or not Get file 0001234A5B6D.cfg … • Solution – Use HTTPS or Encrypted config files
E-Learning Attacks on SIP Phones • How many of you change the default password for IP phones? • How many of you update the IP Phone’s firmware regularly? Video #1Video #2
E-Learning Tips to prevent attacks Easy: 1. Use strong passwords 2. Install and configure iptables 3. Limit SSH to specific addresses or networks 4. Limit HTTP access to specific addresses or networks 5. Change the admin password on phones Hard: 1. Disable the web interface of the phones
E-Learning Damage Control • Face a simple fact, sooner or later, a system open to the Internet will be compromised. • The hacker’s advantage – Administrators have to defend against all attacks, while one vulnerability is enough for the attacker! – The administrator is one, attackers are many!
E-Learning Tips to reduce possible damages? 1. Do not allow all routes to all users. 2. Limit simultaneous calls in International Routes 3. Drop calls after a certain period of time. 4. Use two-way authentication for high-risk international routes Ex. vmauthenticate(mailbox)
E-Learning The absolute minimum Lab. Install iptables and fail2ban
E-Learning • /var/log/asterisk • Controlled by the logger.conf • Log Rotation Where to find logs
E-Learning NoOP Verbose([level,]<message>) CLI>core set verbose 15 CLI>core set debug 6 Logging your own data
E-Learning Logger.conf Tour
E-Learning • Mailing Lists - https://wiki.asterisk.org/wiki/display/AST/Mailing+Lists • Forum - https://community.asterisk.org/ • Wiki http://wiki.asterisk.org Getting help
E-Learning http://issues.asterisk.org
E-Learning • Do not open if: – Information Requests (Use mailing list, IRC or wiki) – Support requests: (Use mailing list, IRC or wiki) – Random wishes and feature requests with no patch – Business development requests (Asterisk-biz mailing list) Opening a support case
E-Learning • Evidences – Do you have strong evidence that it is actually a bug? – Do you have logs and traces? – Are you using one of the supported versions? – All dependencies updated? – Can you reproduce the problem? – Are you available to test the fixes? Before opening
E-Learning • Core Dumps asterisk –g Can produce backtraces essential when your system crashes. #gdb asterisk corefile #/var/lib/asterisk/scripts/ast_coredumper core Debugging
E-Learning • In this chapter you have learned – How to troubleshoot the most common cases – How to enable • Verbose • Debug • Logger – Where to find help – How and when to open a Ticket Summary
E-Learning cdr_custom ;[mappings] ;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${ CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(l astdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CD R(duration)})},${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_ QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)} ; ; High Resolution Time for billsec and duration fields ;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${ CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(l astdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CD R(duration,f)})},${CSV_QUOTE(${CDR(billsec,f)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV _QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)} ;Simple.csv => ${CSV_QUOTE(${EPOCH})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})}
E-Learning Dispositions Value Description NO ANSWER The channel was never answered. CONGESTION Channel congestion. FAILED The channel attempted to dial but the call failed BUSY The remote party was busy. ANSWERED The channel was answered.
E-Learning • Accountcode or Billing-id • Amaflags – Default – Omit, do not record calls – Billing, mark for billing – Documentation mark for documentation • You can set amaflags on the channel – sip.conf, example: amaflags=billing AMAFLAGS and Account Codes
E-Learning • cdr_csv • cdr_custom • cdr_adpative_odbc CDR modules
E-Learning Set(CDR(name)=value) ResetCDR() ForkCDR() CDR associated Functions
E-Learning DEMO – Sending CDRs to a database
E-Learning GotoIfTime GotoIfTime(times,weekdays,mdays,months,[timezone]?[labeliftrue:[labeliffalse]]) Time specification <time range>,<days of week>,<days of month>,<months> <time range>= <hour>’:'<minute>’-‘<hour>’:'<minute>|* <days of week> = <dayname>| <dayname>’-‘<dayname>| “*” <dayname> = “sun” | “mon” | “tue” | “wed” | “thu” | “fri” | “sat” <days of month> = <daynum>| <daynum>’-‘<daynum>| “*” <daynum> = a number, 1 to 31 <hour> = a number, 0 to 23 <minute> = a number, 0 to 59 <months> = <monthname>| <monthname>’-‘<monthname>| “*” <monthname> = “jan” | “feb” | “mar” | “apr” | “may” | “jun” | “jul” | “aug” | “sep” | “oct” | “nov” | “dec”
E-Learning GotoIfTime example GotoIfTime(08:00-18:00,mon-fri,*,*,operator?security);
E-Learning General section variables [general] static=yes ; default = no writeprotect=no ;default=no autofallthrough=yes ;default=yes clearglobalvars=no priorityjumping=yes userscontext=default [globals] CONSOLE=Zap/1 IAXINFO=guest TRUNK=Zap/g2 [default] . Extensions.conf file structure
E-Learning In Depth Look at Applications Applications used in this section  Dial()  Answer()  Hangup()  Background()  Gotoif()  Record()  Playback()  Read()
E-Learning Dial [Syntax] Dial(Technology/Resource[&Technology2/Resource2[&...]][,timeout[,options[,U RL]]]) This application sets the following channel variables: ${DIALEDTIME}: This is the time from dialing a channel until when it is disconnected. ${ANSWEREDTIME}: This is the amount of time for actual call. ${DIALEDPEERNAME}: The name of the outbound channel that answered the call. ${DIALEDPEERNUMBER}: The number that was dialed for the answered outbound channel. ${FORWARDERNAME}: If a call forward occurred, the name of the forwarded channel. ${DIALSTATUS}: This is the status of the call CHANUNAVAIL CONGESTION NOANSWER BUSY ANSWER CANCEL DONTCALL: For the Privacy and Screening Modes. TORTURE: For the Privacy and Screening Modes. INVALIDARGS
E-Learning The most important Dial options D([called][:calling[:progress]]): Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. k or K Allow the called or calling party respectively to enable parking of the call by sending the DTMF sequence defined for call parking in "features.conf". L(x[:y[:z]]): Limits the maximum call time x - Maximum call time, in milliseconds y - Warning time, in milliseconds z - Repeat time, in milliseconds r([tone]): Indicate ringing to the calling party (False Ring) tone - Indicate progress to calling party. Send audio 'tone' from the "indications.conf" tonezone currently in use. S(x): Hang up the call <x> seconds *after* the called party has answered the call.
E-Learning Asterisk Dial options (cont...) t or T allow the called or calling party respectively to transfer the calling party by sending the DTMF sequence defined in "features.conf". This setting does not perform policy enforcement on transfers initiated by other methods. x or X allow the called or calling party respectively to enable recording of the call by sending the DTMF sequence defined for one-touch automixmonitor in "features.conf".
E-Learning Answer() – Answer a ringing channel Progress() – Sends a 183 Session Progress Playback(file,[say,skip,noanswer]) Hangup([causecode]) – Hangup an existing channel Answer, Progress,Playback and Hangup https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings
E-Learning Play an audio file while waiting for digits of an extension to go to. [Description] Background() Options s – skip recording if not answered n - Don't answer the channel before playing the files. m - Only break if a digit hit matches a one-digit extension Background
E-Learning Read() Read() Read a variable [Description] maxdigits -- maximum acceptable number of digits. option  's' to return immediately if the line is not up,  'i' to play filename as an indication tone from your indications.conf  'n' to read digits even if the line is not up attempts -- if greater than 1, that many attempts will be made timeout -- An integer number of seconds to wait for a digit response
E-Learning WaitExten() [aasiptrunk] exten=>9999,1,answer() exten=>9999,n,background(menu1) exten=>9999,n,waitexten(10) exten=>9999,n,set(TIMEOUT(response)=10 exten=>9999,n,Dial(${OPERATOR}) exten=>6000,1,Dial(SIP/zoiper) exten=>6001,1,Dial(SIP/xlite)
E-Learning Record() Record to a file [Description] Options 'a' : append to existing recording rather than replacing 'n' : do not answer, but record anyway if line not yet answered 'q' : quiet (do not play a beep tone) 's' : skip recording if the line is not yet answered 't' : use alternate '*' terminator key (DTMF) instead of default '#' 'x' : ignore all terminator keys (DTMF) and keep recording until hangup 'format' is the format of the file type to be recorded 'silence' is the number of seconds of silence to allow before returning. 'maxduration' is the maximum recording duration in seconds. Record
E-Learning AstDB Family KEY1=VALUE KEY2=VALUE Functions variable=${DB(<family/key>)} DB(<family/key>)=value DB_EXISTS(<family/key>) Applications dbdel(<family/key>) dbdeltree(<family>) CLI Commands database del database put database show <family[/key]> database showkey database deltree database get AstDB
E-Learning Call Forward. BlackList, DND [apps] ;call forward immediate exten=>_*21*XXXX,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten=>_*21*XXXX,2,Hangup exten=>*21*,1,DBdel(CFIM/${CALLERID(num)}) exten=>*21*,2,Hangup ;Do not disturb exten=>_*41*X.,1,Set(DB(dnd/${EXTEN:4})=${EXTEN:4}) exten=>_*41*X.,n,Hangup exten=>*41*,1,DBdel(dnd/${EXTEN:4}) exten=>*41*,2,Hangup ;call forward on busy status exten=>_*61*XXXX,1,Set(DB(CFBS/${CALLERID(num)})=${EXTEN:4}) exten=>_*61*XXXX,2,Hangup exten=>*61*,1,DBdel(CFBS/${CALLERID(num)}) exten=>*61*,2,Hangup AstDB example
E-Learning Call Forward. BlackList, DND AstDB example (Cont)
E-Learning Applications Extensions Language extensions.conf extensions.ael [from-internal] include=>parkedcalls exten => _4.,1,Record(${EXTEN:1}:gsm) exten => _4.,n,wait(1) exten => _4.,n,Playback(${EXTEN:1}) exten => _4.,n,Hangup() exten=>6000,1,gosub(stdexten,(SIP/zoiper)) exten=>_9.,1,dial(SIP/${EXTEN:1}@siptrunk,20) context from-internal { includes { parkedcalls; } =>_4. { Record(${EXTEN:1}:gsm); wait(1); Playback(${EXTEN:1}); } =>_6000 { gosub(stdexten(SIP/zoiper)); } =>_9. { Dial(SIP/${EXTEN:1}@siptrunk,20); }
E-Learning Limiting simultaneous calls exten=>_214X,1,set(GROUP()=Rio) exten=>_214X,n,Gotoif($[${GROUP_COUNT()} > 1]?overflow) exten=>_214X,n,Dial(SIP/${EXTEN}) exten=>_214X,n,hangup exten=>_214X,n(overflow),playback(exceededcalls) exten=>_214X,n,hangup
E-Learning • In this section you have learned some of the most important concepts in the dialplan. – Context inclusion – Dialplan processing order – In depth look at some of the main applications – Asterisk DB – Application Extensions Language – Grouping and limiting calls Section summary
E-Learning 1. res_pjsip – The main services and the base layer 2. res_pjsip_session – Media session and addons 3. res_pjsip_messaging – text messages 4. res_pjsip_registrar – registrations 5. res_pjsip_pubsub – subscribe/notify/publish PJSIP modules
E-Learning 1. Endpoints (primary object) 2. AOR (Address of Record) 3. Auth - authentication 4. Transport 5. Registration 6. Identify (By User, IP..., equivalent to peer matching) Objects defined in PJSIP
E-Learning PJSIP entity relationship model Source: wiki.asterisk.org https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships
E-Learning Transport configuration [transport-udp-main] type=transport protocol=udp bind=0.0.0.0:5060 [transport-udp-alternative] type=transport protocol=udp bind=0.0.0.0:5080
E-Learning [xlite] type=endpoint transport=transport-udp-main context=from-internal disallow=all allow=ulaw aors=xlite auth=xlite [xlite] type=auth auth_type=userpass username=xlite password=#supersecret# [xlite] type=aor max_contacts=1 SIP phone configuration
E-Learning SIP trunk configuration [siptrunk] type=endpoint transport=transport-udp-main context=from-siptrunk direct_media=no disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk [siptrunk] type=aor contact=sip:sip.api4com.com:5600 [siptrunk] type=auth auth_type=userpass username=1020 password=supersecret [siptrunk] type=registration outbound_auth=siptrunk server_uri=sip:1020@sip.api4com.com:5 600 client_uri=sip:1020@sip.api4com.com contact_user=9999 [siptrunk] type=identify endpoint=siptrunk match=sip.api4com.com
E-Learning Converting sip.conf on pjsip.conf sip_to_pjsip.py [options] [input-file [output-file]]
E-Learning • pjsip console commands tour PJSIP console commands
E-Learning • Helpful to troubleshoot SIP • Excelent article at https://blogs.asterisk.org/2016/02/24/debugging- sip-message-traffic-with-pjsip-history/ PJSIP History
E-Learning exten => _6XXX,1,Dial(PJSIP/${EXTEN}) exten => _6XXX,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}) exten => _9.,1,Dial(PJSIP/siptrunk/sip:${EXTEN:1}@sip.api4com.com) exten => _9.,1,Dial(PJSIP/${EXTEN:1}@siptrunk) PJSIP naming conventions
E-Learning Access Control Lists ACLs ; Define the ACL in pjsip.conf or acl.conf [acl] type=acl deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/24 deny=192.168.0.1 ; The ACL configuration is independent of individual endpoint configuration and ; operates on all inbound SIP communication using res_pjsip.
E-Learning • PJSIP is more powerful – Multiple AORs and Contacts, parallel forking – Multiple transports – Easier to extend – History and Channel Stats • PJSIP is not simple to use – Too many objects and sections – Adoption rate is still very slow – Less robust, due the lack of a large user base • No significant performance differences Comparing channels
E-Learning • Converting to PJSIP LAB 5
E-Learning Additional information on chan_sip
E-Learning • By the end of this chapter you should be able to – Construct a SIP channel acording to the naming conventions – Understand how peers are matched for incoming calls – Apply the most useful CLI commands to troubleshoot SIP – Enable SIP tracing with SIP set debug – Configure rtptimeout – Configure QoS tagging Objectives
E-Learning • Friend – Entity to send and receive calls • Peer – Entity to send calls • User – Entity to receive calls • Host=dynamic, means it can register Peer Types
E-Learning Peer Matching A TYPE=USER SECTION EXISTS AND MATCHES THE “FROM” HEADER FI ELD? CHECK “FROM”HEADER FIELD IN THE SIP PACKET AGAINST THE SIP.CONF YE S A TYPE=PEER SECTION EXISTS AND MATCHES THE IP ADRESS DEFINED IN HOSTS=IP INSECURE=IN VITE? YES ALLOWGUEST ? TRUE ALLOW USE CONTEXT FROM GENERAL SECTION FALSE DENY NO NO MD5 CREDENTIALS ALLOW/DENY ALLOW USE CONTEXT FROM THE “USER” SECTION OK DENY NOT OK NO MD5 CREDENTIALS ALLOW USE CONTEXT FROM PEER SECTION OK DENY NOT OK YE S ALLOW USE CONTEXT FROM PEER SECTION
E-Learning Channel Naming SIP/[exten@]peer[:portno] exten => s,1,Dial(SIP/ipphone) exten => s,1,Dial(SIP/info@voffice.com.br) exten => s,1,Dial(SIP/192.168.1.8:5060,20) exten => s,1,Dial(SIP/8500@sip.com:9876) exten => s,1,Dial(SIP/username:password@sip.com:9876/8500)
E-Learning Console Commands ; CLI Commands ; ------------------------------------------------------------- ; Useful CLI commands to check peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show registry Show status of hosts we register with ; ; sip set debug on Show all SIP messages ; ; sip reload Reload configuration file ; sip show settings Show the current channel configuration ;
E-Learning SIP debug • sip det debug on to enable • sip set debug off to disable
E-Learning RTP timeout ; The settings are settable in ; the global section as well as per device ; ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity ; on the audio channel ; when we're not on hold. This is to be able to hangup ; a call in the case of a phone disappearing from the net, ; like a powerloss or grandma tripping over a cable. ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity ; on the audio channel ; when we're on hold (must be > rtptimeout) ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero)
E-Learning Configure QoS tagging ;tos_sip=cs3 ; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. ;tos_text=af41 ; Sets TOS for RTP text packets. ;cos_sip=3 ; Sets 802.1p priority for SIP packets. ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets. ;cos_video=4 ; Sets 802.1p priority for RTP video packets. ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
E-Learning NAT on client chan_sip qualify=yes directmedia=no directmedia=nonat nat=force_rport,comedia nat = no ; Do no special NAT handling other than RFC3581 nat = force_rport ; Pretend there was an rport parameter even if there wasn't nat = comedia ; Send media to the port Asterisk received it from regardless ; of where the SDP says to send it. nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default) nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
E-Learning NAT on client PJSIP [phone] type=endpoint direct_media=no rtp_symmetric=yes force_rport=yes [phone] type=aor qualify_frequency=15
E-Learning Server behind NAT on chan_sip externaddr = 12.34.56.78 ; use this address. localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet=169.254.0.0/255.255.0.0 ; Zero conf local network externhost=foo.dyndns.net ; refreshed periodically externrefresh=180 ; change the refresh interval
E-Learning Server behind NAT on PJSIP [transport-udp-nat] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.0.0/16 external_media_address=203.0.113.1 external_signaling_address=203.0.113.1 external_signaling_port=5060
E-Learning Application Layer Gateway • Solves the NAT problem in the router Workarounds: Disable ALG in the router Use a port diferent than 5060
E-Learning Section Summary • SIP • PJSIP • Advanced Parameters of CHAN_SIP • NAT Traversal • NAT Traversal on Client and Server

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