Asterisk Support - Inspired by Asterisk Pro Installers.( API )

Index
Asterisk Hardware
Suppliers
The Installers
zaptel.conf
zapata.conf
sip.conf
iax.conf
extensions.conf
Problem Solving

Asterisk Hardware
On the Trunk side :

Analogue Trunks

Diguim TDM cards with Digium or Openvox FXO modules

PRO's : Inexpensive , High-density cards are available.

CON's : Watch out for the cards without Echo-cancellation!...cancel%$on...can#&%..

Openvox TDM cards with Openvox or Digium FXO modules

PRO/CON's : Same as above - try to buy the cards made-up , buying modules loose get's expensive.Try to get the high-density cards if you are planning to grow in the future.

FXO Gateways from Micronet,Planet,Sipura etc

PRO's : Don't need available PCI slots and can be used where 2U 19" rack mounted systems can not accomodate TDM cards. Also uses SIP.

CON's : Generally more expensive and a network overhead

Digital Trunks (ISDN)

ISDN Basic Rate ( BRI )

BRI cards from Junghanns - up to 8 BRI's on 1 PCI card

PRO's : These cards really work well with EURO ISDN !. Use the bri-stuff drivers.

CON's : Can't use 2 Quad cards for 8 BRI's! So watch out if you are planning for growth!

BRI modems from Duxbury for a single BRI

PRO's : At under R300 who can say no.

CON's : Sticky but works with mISDN/vISDN/ZAP-HFC.

ISDN Primary rate ( PRI ) or E1

PRI cards from Sangoma - up to 4 PRI's on 1 PCI card

PRO's : These cards really work well with EURO ISDN !. Use Wanrouter drivers.

CON's : Like I said they really work well.

PRI cards from Digium

PRO's : They are avilable.

On the Extension side :

VOIP Handsets from Linksys,SNOM,Sipura and Aastra.

PRO's : Feature rich , LCD display , programmable keys , hey it's the way all phones will be made someday.

CON's : The bottom line remains - expect to get what you pay for!

VOIP ATA (Analogue Gateways) from Linksys,Sipura and Digium.

PRO's : Get 2 channels for the price of one.External so no PC limitations.

CON's : No callerid , NO LCD display no programmable keys ... You can get aound these problems with FOP or YAACID for example , but it's not the same.
TDM cards with FXS modules from Digium and/or Openvox.

PRO's : Easy to install , relatively inexpensive compared with VOIP handsets.

CON's : Same problems as far as features and info is concerned.Uses PC resources.

Index

  • Suppliers :

    • Evenflow

    Evenflow is great to work with and sells everything from TDM cards to VOIP phones.

    Dow Networks

    Dow Networks supports IAX2 Trunking and is therefore our ITSP of choice

    Scoop Distribution

    Scoop's support is great and does the Atcom Range - TDM cards and VOIP phones.

    Connection Telecom

    Connection Telecoms is the Official Channel partner for Digium in South Africa.

    Patton
    Voipmagic sells the Patton range of VOIP products.

    Index

  • The Asterisk Pro Installers

    • Delegates attending the course range from Entrepreneurs to Call Centre Managers to the Sales force of PBX companies - the Feedback is extremely positive and we are proud to say that we have contribited in a big way to the understanding of asterisk.

    These are very special , forward thinking individuals and I believe the companies and businesses that employ or support them are very lucky indeed !

    Index

  • The zaptel.conf
    • Consider zaptel.conf as the driver-instruction file - in other words , zaptel drivers work with zaptel hardware like PSTN interfacing cards.Your zaptel.conf tells linux how many fxo/fxs modules are on your card and what signalling you would like to use for example.It is important to remember to use the opposite signalling to the module that's installed - in other words add fxs signalling when installing FXO modules & visa versa.
    Simply put the following in your zaptel.conf :

    fxsks = 1 ; assuming you have 1 FXO moduel installed on the 1st port.

    What you are looking at is the very basics of zaptel's configuration , alot can be added to streamline and clean the performance of your hardware , but this should get you started.

    The config files will look very different when using ISDN !

    * AN IMPORTANT NOTE : DO NOT PLUG A LIVE TRUNK INTO A FXS PORT - YOU MIGHT CAUSE IRRIVERSABLE DAMAGE TO THE HARDWARE !!!


    Index

  • The zapata.conf
    • Like all the files in your /etc/asterisk/ directory these control the way asterisk deals with these modules.In the case of zapata.conf it further deals with the way your zaptel hardware will be used within your asterisk dialplan. Important things to add are the following :
    [channels]

    signalling = fxs_ks ; correspond with zaptel.conf , notice the underscore!

    channel => 1 ; correspond with zaptel.conf

    context = incoming ; has to exist in your dialplan!


    Index

  • The sip.conf
    • Sip.conf is where your registration information for all your SIP clients are kept. You can also add feature-rich sip info to each client and seperate them by assosiating them with different contexts.
    Now program your SIP phone via it's web-interface to match these settings and point to your asterisk server as it's sip-server

    Simply put the following in sip.conf

    [general]

    bindport=5060

    [100] ; make username/account 100 on your sip phone

    context = internal ; must exist in your dialplan!

    secret = 100 ; match password in phone

    disallow = all ; start by disallowing all codecs

    allow = ulaw ; ulaw = G711a

    allow = alaw ; alaw = G711u

    dtmfmode = info ; Set default dtmfmode for sending DTMF.

    host = dynamic ; Accept all ip addresses

    type = friend ; Options : friend/user/peer

    mailbox = 100@default ; Add user 100 in voicemail.conf


    Index

  • The iax.conf
    • What will happen here is that you have a * server with username : local_server - connecting to a remote * server with a user called remote_server.You will have these settings below , and the remote server will have exactly the same except that the [contexts] will be swopped around and the remote server will have your ip address and you will have the remote server's ip address in your peer settings.In other words YOU will be a PEER of the remote server and the remote server will be a PEER of your server.With connection via the internet you will probably use dynamic domain names ( example : remote.dyndns.org ) if you don't have fixed public ip's.
    Remember to do the neccesary port forwarding ! ( port 4569 with iax ) on the router

    Simply put the following in iax.conf

    [general]

    bind port = 4569

    disallow = all

    allow = ulaw

    allow = gsm

    [local_server]

    type = user

    auth = md5

    host = dynamic

    secret = password for remote server

    context = internal ; must exist in your dialplan!

    [remote_server]

    type = peer

    auth = md5

    host = ipaddress of remote server.

    secret = this password MUST match the remote server USER password.


    Index

  • The extensions.conf
    • The absolute Heart of asterisk - also affectionately known as the Dialplan. This is where all the instructions is kept for interacting with asterisk.
    To test your extension , you can use the Echo() Application - this will simply echo whatever you say - hence the "Echo"

    Simply add this to your extensions.conf

    [internal]

    exten => 123,1,Answer()

    exten => 123,2,Echo()

    Now when you dial 123 , asterisk will answer and echo your words !

    You are going to create extension 100 in order for users to dial SIP-Phone-User 100

    Simply add this to your extensions.conf under your existing code

    exten => 100,1,Dial(SIP/100)

    exten => 100,2,Congestion()

    And now create extension 101 in order for users to dial SIP-Phone-User 101

    Simply add this to your extensions.conf under your existing code

    exten => 101,1,Dial(SIP/101)

    exten => 101,2,Congestion()

    Now send all incmoing calls from the PSTN (connected to zaptel hardware) to your SIP phone

    Add these line to your extensions.conf under your existing code

    [incoming]

    exten => s,1,Dial(SIP/100)

    exten => s,2,Busy()


    Index

  • Problem Solving!
    • This is the most important and probably the most difficult part of asterisk without a doubt! Use your console ( the CLI> ) to access live info from asterisk. Either use the asterisk -rvvvvvvv command from the shell or enter " set verbose 7 " in the console. Asterisk also creates a log file - be careful not to allow your logfile to become to big - there are known problems with lof files that becomes too big ( >10M ) and causing all sorts of problems - so remember to turn your verbosity level back to zero when done debugging!

    COMMON PROBLEMS & HICKUPS :
    The 3 most common mistakes are 1) Typo's 2) Typo's and 3) YES Typo's !!! -so go and look for spelling and syntax mistakes 1st.

  • The digits doesn't get read when accessing Voicemail / IVR ?
    • - Check that your "dtmfmode=" in sip.conf matches your sip phone settings.

  • Asterisk doesn't release the PSTN trunk after a hangup.
    • - Add the following lines to zapata.conf :
    ;For Cape Town Exchanges

    busydetect = yes

    busycount = 4

    busypattern = 500,500

    callprogress = no

    ;For the Rest of the country's Exchanges

    busydetect = yes

    busycount = 2

    busypattern = 2500,500

    callprogress = no

    Also change this setting : country = za in indications.conf and comment/uncomment for the correct exchange under the [za] context.

  • Asterisk doesn't want to start at all.
    • - Try starting with # asterisk -cvvvvv If you see the zaptel module errors - your zaptel & zapata config doesn't match your hardware & * won't start.

  • There is alot of echo...echo....echo...
    • - Try using the fxotune application to finetune your FXO modules.
    Modify /etc/asterisk/zaptel-*.*/zconfig.h as follow :

    You need to edit the zconfig.h source file and comment out the line #define ECHO_CAN_KB1 and uncomment #define ECHO_CAN_MARK2 on about line 52. You also need to enable aggressive echo cancellation by uncommeting #define AGGRESSIVE_SUPPRESSOR

    Modify /etc/modprobe.conf as follow:

    install wctdm /sbin/modprobe –ignore-install wctdm opermode=SOUTHAFRICA && /sbin/ztcfg

    You need to edit the zaptel.conf file : Change defaultzone=us, loadzone=us to defaultzone=za, loadzone=za

  • I need to log the calls from asterisk and bill them correctly.
    • - Asterisk is logging the calls by default to the Master.csv file in the /var/log/asterisk/cdr-csv/ folder , all you need is something like Man3000 and send the data to the COM port - simply use the tail command and send it to /dev/ttys0 for COM 1 , then program your Man3000 software to grab the correct data. You can further manipulate your output by changing the [mappings] in the cdr_custom.conf file.


    Index

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