Administration Guide

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Administration Guide
This is a book-style Wiki (or a Wiki-style book) that will become complete Administrators Guide to FreePBX. To help, add a child page to this page, writing a section for each of the major items in the rough outline. Pick whatever you like. If it's not one of the categories below, or belong to them, think carefully if it belongs here at all. It may be more useful someplace else. Rough outline: Installation 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11. 12. 13. 14. 15. 16. Information gathering Putting the system together Starting from a blank slate Creating and assigning extensions. Setting up voicemail Creating an IVR. Creating Queues. Setting up backup and restore User control: How to let the user at a little bit... User Portals and the ARI Training New Users on how the system is configured Transitioning to the new system Running a help desk using voice, tickets, and email Connecting POTS lines Connecting PRI trunks How to connect VOIP trunks 1. How to test a new IP line for VOIP quality 2. Two-way trunks 3. One-way trunks 17. Outbound routing 18. Inbound routing Administration 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11. 12. Moves, adds, changes, and deletes: How to administer extensions with a minimum of pain. Creating, changing and deleting IVRs. Creating, changing and deleting Queues. Backup and restore: From cron to Oh, No! User control: How to let the user at a little bit... Training New Users on how the system is configured New Equipment: How to add with a minimum of disruption Upgrades: How and when to do it. Running a help desk using voice, tickets, and email How to move a PRI How to move a VOIP trunk. How to test a new IP line for VOIP quality


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13. How to mix VOIP and data on the same LAN 14. How to mix VOIP and data on the same backhaul

Adding Extensions Adding Extensions
A PBX without any extensions isn't very useful, so it's the first thing to do after installing FreePBX. Extensions let you test all kinds of things, so it's the first thing to get right.

Shown at right are a few test extensions on a FreePBX installation on my t42 Ubuntu laptop.

There are several pages of information here. We'll go through each of them.

Display Name: This is the name that is used, at least internally, when placing an outbound call. Most Caller Name services look up the name in a database, so this name setting might do nothing on your outbound VOIP or PRI calls. It will certainly do nothing on outbound POTS calls. CID Num Alias: The CallerID to show when dialing intracompany. Example Usage: James has a office extension at 201, a softphone at 401, a home office phone at 601, and a FollowMe at 201 that rings them all. 401 and 601 can use a CID Num Alias of 201, so that all internal call recipients see “201” SIP Alias: Every 'clever' presentation of VOIP has an example of dialing by email address. This is hard to do on most phones, but is nonetheless supported. Put only the name here, not the @ symbol or the


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fully-qualified-domain name. That's used by the calling application or device to locate your PBX on the Internet. To allow any party to call you, you'll need to have firewall rules that allow all SIP calls regardless of IP address. This is only advisable if your Asterisk installation is up-to-date, and has no current SIP security vulnerability. Direct DID: This is where you enter the Direct Inward Dial (DID) you'd like to reach this extension. If you forget, all calls to that DID will end up at the main IVR. Putting a value here eliminates the need to create an Inbound Route. DID Alert Info: Used for distinctive ring services

Music on Hold: Set a different Music On Hold (MOH) class for this extension. Great for having different music for different offices or companies that are served by the same PBX. Outbound CID: Put the CallerID and preferred CallerIDName here for outbound usage. Ring Time: How long to ring before a server-side transfer to voicemail. You'll usually use the default here, and set a system-wide value in General Settings. Call Waiting: Set the call waiting value. Also accessible by feature code from an individual extension (by default *70 to activate and *71 to deactivate – see Feature Codes). Emergency CID: The CallerID to be set when dialing a number labeled as emergency.

Device Options
Extensions - Device Options These options are the same as in a vanilla asterisk sip.conf file. In a FreePBX installation, they end up in sip_additional.conf. For more information, check out Asterisk: TFOT. secret: The SIP password used in the authentication of this device to the server. dtmfmode: How DTMF is expected by the server. Options are rfc2833, INFO, and in-band. rfc2833 seems the most reliable across many devices. Client devices (e.g. Linksys) often have an Auto setting, which is to be avoided.


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canreinvite: Asterisk is a back-to-back useragent. This means that your phone calls it, and it calls your VOIP, PRI or POTs line. All audio (RTP stream) is carried through the Asterisk process during the call. Your VOIP service provider, for example, often will use a SIP REINVITE message to change the RTP destinations after the call is set up. This reduces load on the equipment, as it's only doing call setup and takedown. Highly desirable if you're supporting remote users making VOIP calls and your VOIP provider supports REINVITE. However, it's tricky to get any of your FreePBX features to work in this scenario. Play with this, but don't use it on a customer system unless you have tested the features you need. context: Context is an Asterisk dialplan sphere-of-influence concept used to separate components from each other (multi-tenant, for example, or outward facing customer service from backoffice). From-internal means you can dial like you're a phone on premises with access to other extensions and outbound trunks. Other common options are outbound-all-routes (dial out only), from-trunk (extensions only, no outbound dialing) host: dynamic or a static IP address. dynamic allows any device that can pass the SIP challenge/authentication to register and make/receive calls. type: friend or peer. Use friend for a phone. Peer is for SIP devices that are capable of carrying calls, like a Trunk. nat: yes or never. SIP is a nat-unfriendly protocol in that it specifies the return IP address for the call audio stream deep inside a packet. NAT works by rewriting packet source and destination IP addresses, but doesn't understand SIP (unless a good SIP Application Layer Gateway is installed). NAT is therefore problem if both the phone and the server PBX are separated from the public Internet by different NATs (e.g. a home router and and corporate one.) In such a situation, audio won't work, but signaling will (phones will ring but no audio). To support remote home users behind conventional NATs, use yes, and either give the server PBX a public IP address or do a 1:1 IP mapping from a public IP to it's internal, then set IP_nat.conf to the public IP address of the system. NAT=yes instructs Asterisk to send audio to the IP it receives it from, regardless of what the SIP SDP says, and lets you have at least one NAT present and still have effective audio. Note that NATs vary widely as to how long they stay 'open'. Best practice when using Non-STUN phones is to have SIP registration expire every 60 seconds – the re-registration (outbound, by the phone) will keep the NAT open to receive calls. NAT=yes doesn't hurt anything when the client device is on the same LAN. callgroup: pickupgroup: disallow: enter codec overrides here. An extension or group of extensions on a low-bandwidth link might want to disallow the higher-bandwidth codecs out of the general pool. allow: enter any codec overrides here dial: SIP/extension is the default.


freepbx. or specific help on a command include context: Include context in other context load: Load a dynamic module by name logger reload: Reopen log 4/20/2011 . debug channel: Enable debugging on a channel dont include: Remove a specified include from context help: Display help list. Use after rotating the log files. Asterisk CLI Commands General commands !<command>: Executes a given shell command abort halt: Cancel a running halt add extension: Add new extension into context add ignorepat: Add new ignore pattern add indication: Add the given indication to the country amportal start: Stop AAH and amportal stop: Restart AAH.Administration Guide Page 5 of 42 accountcode: enter an account code for use by a billing module. mailbox: extension@default is the default. no debug channel: Disable debugging on a channel pri debug span: Enables PRI debugging on a span pri intense debug span: Enables REALLY INTENSE PRI debugging pri no debug span: Disables PRI debugging on a span remove extension: Remove a specified extension remove ignorepat: Remove ignore pattern from context remove indication: Remove the given indication from the country save dialplan: Overwrites your current http://www.

2 AGI Commands http://www.conf file with an exported version based on the current state of the dialplan.conf is not saved. the current values of global variables are not written into the new extensions. set verbose: Set level of verboseness show agents: Show status of agents show applications: Shows registered applications show application: Describe a specific application show channel: Display information on a specific channel show channels: Display information on channels show codecs: Display information on codecs show conferences: Show status of conferences show dialplan: Show dialplan show image formats: Displays image formats show indications: .2.freepbx. Using "save dialplan" will result in losing any comments in your current extensions.conf. A backup copy of your old extensions. The initial values of global variables defined in the [globals] category retain their previous initial values.Administration Guide Page 6 of 42 extensions.conf.Show a list of all country/indications show locals: Show status of local channels show manager command: Show manager commands show manager connect: Show connected manager users show parkedcalls: Lists parked calls show queues: Show status of queues show switches: Show alternative switches show translation: Display translation matrix show voicemail users: List defined voicemail boxes show voicemail zones: List zone message formats soft hangup: Request a hangup on a given channel 4/20/2011 .

2.Administration Guide Page 7 of 42 show agi: Show AGI commands or specific help dump agihtml: Dumps a list of agi command in html format 4/20/2011 .4 IAX Channel Commands iax2 debug: Enable IAX debugging iax2 no debug: Disable IAX debugging iax2 set jitter: Sets IAX jitter buffer iax2 show cache: Display IAX cached dialplan iax2 show channels: Show active IAX channels iax2 show peers: Show defined IAX peers iax2 show registry: Show IAX registration status iax2 show stats: Display IAX statistics iax2 show users: Show defined IAX users iax2 trunk debug: Request IAX trunk debug iax debug: Enable IAX debugging iax no debug: Disable IAX debugging iax set jitter: Sets IAX jitter buffer iax show cache: Display IAX cached dialplan iax show channels: Show active IAX channels iax show peers: Show defined IAX peers http://www.3 Database Handling database del: Removes database key/value database deltree: Removes database keytree/values database get: Gets database value database put: Adds/updates database value database show: Shows database contents A.freepbx. 4/20/2011 .Administration Guide Page 8 of 42 iax show registry: Show IAX registration status iax show stats: Display IAX statistics iax show users: Show defined IAX users init keys: Initialize RSA key passcodes show keys: Displays RSA key information A.1 on 2004-01-23) sip show channels: Show active SIP channels sip show channel: Show detailed SIP channel info sip show inuse: List all inuse/limit sip show peers: Show defined SIP peers (register clients) sip show registry: Show SIP registration status (when Asterisk registers as a client to a SIP Proxy) sip show users: Show defined SIP users A.6 Server management restart gracefully: Restart Asterisk gracefully restart now: Restart Asterisk immediately restart when convenient: Restart Asterisk at empty call volume reload: Reload configuration stop gracefully: Gracefully shut down Asterisk stop now: Shut down Asterisk immediately stop when convenient: Shut down Asterisk at empty call volume extensions reload?: Reload extensions ONLY unload: Unload a dynamic module by name show modules: List modules and info about them http://www.2.7.5 SIP Channel commands sip debug: Enable SIP debugging sip no debug: Disable SIP debugging sip reload: Reload sip.2.conf (added after 0.

I gave a common password xxxyyy to both boxes. you will probably be spending a good part of 3 hours trying to get them to talk to one another. but I prefer 2 separate extensions as I have them working). to set 2 very basic systems together (you can refer to DUNDi for a more complete solution).au) together and if you are like me. the Main Office ( and xyz. I settled for the simplest solution and after some fiddling around I managed to get them to work the way I wanted it but not happy with it.freepbx. Trunk Name Parramatta 4/20/2011 . I will just create a few tables outlining what I did.1 METHOD 1 . Instead of being verbose in my explanation.with the peer Asterisk boxes as extensions For the purpose of registering the peers to each other. I hope this will help those in the same position as I I have 2 different locations.Administration Guide Page 9 of 42 show uptime: Show uptime information show version: Display Asterisk version info Connecting 2 or more boxes There may be a time when you want to interconnect 2 Asterisks boxes (def. with about 11 extensions and another office in a different location (xyz. The main office is the only box that will have accounts with different VSPs and all external communications are through the main office Asterisk about 20 km away with 9 extensions. Avoid using extension starting with 8 as it may clash with conferencing. I created 1 extension on each box eg: 90000 on System 1 and 91000 on System 2– using extension numbers that I am not likely to use as local extensions (while some users have had success using common extension. I solicited some advise from a friend (thanks to Mark Brooker) who told me that my configuration could be made a lot tidier. For simplicity. That I did. System 1 System 2 IAX Trunk Outgoing Dial Rules: XX. 90000:xxxyyy@def.Administration Guide Page 10 of 42 MainOffice Peer Details 4/20/2011 .au Note: Registration isn’t really It will still work without it unless you use Dynamic IP. System 1 System 2 Extensions Phone Protocol IAX IAX Extension Number 90000 91000 Extension Password (or IP) secret=xxxyyy type=peer username=90000 User Context Leave blank Leave blank User Details Leave blank Leave blank Register String (or IP) secret=xxxyyy type=peer username=91000 host=def.

for system 1 and system 2 respectively instead of just 6XXX and XX.(Apart from Local extensions. you may place a prefix 4/20/2011 . as it requires an extension to be created for the peer Asterisk box. it will not pass the calling party extension number to the remote Asterisk box.Administration Guide Page 11 of 42 xxxyyy xxxyyy Fullname Parramatta Main Office Voicemail & Directory Disabled Disabled System 1 System 2 Outbound Routing Route Name Parramatta MainOffice Route Password Leave Blank Leave Blank Dial Patterns 6XXX(6001 to 6009 are Parramatta Office extensions) XX. you will need to register both the boxes with DynDns to obtain a valid DNS ID. 9|6XXX and 9|XX. The above example assumes that both Asterisk boxes have Public Fix IP address. If you have Dynamic IP addresses. Note: While this method will provide some rudimentary security (though pretty weak).freepbx. If you want to use a prefix to dial the remote extensions and to use the remote routing rules. Instead.g. If you are a part of a Corporate LAN. all others go via City Office) Trunk Sequence IAX2/Parramatta IAX2/MainOffice The above Outbound Routing rule assumes that you do not wish to use a dialling prefix. than you will have no need to worry about DynDns and what not. it will http://www.

as this is pretty wide This method treats both the Asterisk box as internal to each other as peer and user. In many ways. As different installation resorts to different types of security arrangement. This method does not require registration either and does not require you to create extensions for the peers.2 METHOD 2 . however I believe. you must provide for security. I will leave that to the individual implementer to deal with the security (or IP) Qualify=yes http://www. I am using IAX2 for this purpose. Note: You must provide for security. Unlike the first method. Rather than being verbose.Administration Guide Page 12 of 42 pass the Trunk ID only and all calls will seem to come from the same trunk and not individual extension – I did say that this is a simple solution. Like all installation. I will illustrate this method using tables as follows. System 1 System 2 IAX2 Trunk Outgoing Dial Rules: 6XXX XX. this second method will pass the Caller ID to the receiving party.freepbx. this is simpler to set up. The receiving party will actually get the callers’ extension number/ID instead of the extension number of the peer Asterisk box. 4/20/2011 . Trunk Name InterOffice InterOffice Peer Details host=xyz. (Note: A little tutorial on DUNDi can be found here). you may be able to do this with SIP as well if you are trying to connect the older Asterisk with the newer incarnations (I have not proved it yet).In a Peer/User arrangement Another method that I use is described below.

com.Administration Guide Page 13 of 42 type=peer host=def.(Apart from Local 4/20/2011 .com. all others go via City Office) Trunk Sequence IAX2/InterOffice IAX2/InterOffice Thinking of more than 2 boxes? (or IP) type=user context=from-internal (or IP) Qualify=yes type=peer User Context InterOffice-In InterOffice-In User Details context=from-internal (or IP) type=user System 1 System 2 Outbound Routing Route Name InterOffice InterOffice Route Password Leave Blank Leave Blank Dial Patterns 6XXX(6001 to 6009 are Parramatta Office extensions) XX.

both Apache and iptables can be used to restrict access on a location basis to the web application. – with the appropriate dial plan of course. Box A is the master box. Both the above 4/20/2011 . This lets you give office managers. While I have connected 3 boxes successfully. while useable for a basic configuration.B peers with A . http://www. will not provide you with a complete solution. To provide a complete solution is beyond the scope of this document. Mie is allowed to see status. my advise to you is to hire a VOIP consultant. A peers with B and C . The following link will provide further reference for connecting two Asterisk boxes together http://www.I believe. Creating Administrator Roles Creating Administrator Roles For most web applications it is useful to have graduated permission access. B and C (System If you require a complete solution tailored to your exact requirement. edit extensions (this part is not shown) and apply changes. for example. All the other boxes use box A as the main exchange. without exposing trunks and other settings they do not need. Show below is just such a configuration.Administration Guide Page 14 of 42 Just as a matter of interest.voip-info.freepbx. In addition to the webapp username / password settings. In my implementation I have box A. se same principle can be applied to more boxes.And C peers with A. all external and inter-office (inter-branch) traffic goes via Box A. though exceptions can be be made for the Recordings (ARI) interface. you can connect several boxes using this method. A good policy is to only allow local (LAN) or tunneled via SSH access to the web application. 2 and 3). In this case. so that users have only access to the functions they need. Except for local traffic. access to the Extensions directory to change usernames and reset voicemail passwords as employees come and go.

and then make the inevitable changes. day/night modes or time conditions). etc. Write out word-for-word what all the recordings are going to be. press 5 for office directions. Hospitality 1. Press 1 for sales. or press 0 for the operator. Welcome to BUSINESSNAME. 6.Administration Guide Page 15 of 42 Creating an IVR Digital Receptionist or IVR Information The 'Digital Receptionist' page is the interface used to setup your auto attendant when people call your 4/20/2011 . 3. Bask in glory! Standard IVR Examples: 1. First. for Service press 2". Now upgrade the voice prompts to a paid voice or designated employee (the office manager or receptionist. The proper flow to build a good IVR is: 1. 5. Test all of these. If you know the extension of the person you are trying to reach.) # to access the company directory. 7. Record the audio prompts using System Recordings and an extension. One way to do this is use miscellaneous destinations. draw out on paper what you intend to to achieve. Then go create your IVR. Press 1 http://www. Welcome to HOTELNAME. Planning While the urge is strong just to dive in by clicking on IVR. press 4 for Press inquiries. 2. 8. If you know the room # of the guest you are trying to reach. Create any destinations that don't currently exist (queues. 4. you may dial it at any time. press 3 for administration. Normally heard as "Thanks you for calling MYBUSINESS. ring groups. press 2 for customer service. Planning 2. you may dial it at any time.freepbx. Show it to the customer. Please listen carefully as our options have changed. etc. assigning a * feature code to whatever thing you want to test. Office / Light industrial 1. Please listen carefully as our options have changed. you should resist this impulse. for Sales press 1. Customer agreement with the plan. Run it by the customer (or your officemates).

Administration Guide Page 16 of 42 for reservations. press 5 for Press inquiries. locations. or press 0 for the operator. you may dial it at any time. press 3 for technical support. Engineering/Product Company with Direct Sales and Support 1. Please listen carefully as our options have changed. press 2 for the front desk. Making recordings Fire up the System Recordings module. press 2 for customer service. If you know the extension of the person you are trying to reach. press 4 for administration. press 4 for administration. Press 1 for sales. press 2 for customer service. press 3 for event sales. Please listen carefully as our options have changed. and directions. press 4 for hotel administration. I strongly suggest you use an extension connected to the PBX to make your recordings. Retail 4/20/2011 . or press 0 for the operator. Welcome to BUSINESSNAME. They'll be quick http://www. or press 0 for the operator.5. press 5 for Press inquiries. 3. Press 1 for sales.freepbx. press 6 for office directions. press # to access the company directory. If you know the extension of the person you are trying to reach.1. Shown here is 3. press 5 for hotel directions.3. 4. you may dial it at any time. press 3 for store hours. Welcome to BUSINESSNAME. press # to access the company directory. press # to access the hotel directory.

You can either edit an IVR. You can listen to your recording and add on other recordings (such as the built-in recordings) by clicking on your recording in the right tool panel. we can create our IVR. Now that we've created a system recording. enter your extension in Step 1 and press Go. Dial *99 to listen to 4/20/2011 . the first page is now a brief set of instructions on how to drive the IVR. If the recording is good enough (and don't obsess here yet). spaces are not allowed in the names. you can come back and replace those temporary recordings with paid or improved versions. You don't have to be the person doing this – I often enter a customer's extension and have a customer do this part while I do the GUI work.Administration Guide Page 17 of 42 and in the right format and you can worry about getting everything else right. Don't skip this and go to Step 2. When everything is all finished.freepbx. Now dial *77 and make your recording after the beep. Creating the IVR When you select IVR. or create a new one by clicking on 'Add IVR'. For lame and silly reasons. name the recording and press Save. or you'll get a cryptic error. so the single Welcome-to-ACME recording will be enough. To use your extension to make a recording. if one is existing. Editing your IVR http://www. We're going to start with a simple 1-level IVR .

this creates the IVR (and calls it 'Unnamed') as soon as you click 'Add' . or a series of numbers. Announcement: A System Recording that is played to users when they enter the IVR. enter the option for the user. and in the dropdown menu of Destinations Timeout: This is the amount of time the system waits before sending the call to the 't' destination Enable Directory: If you switch this on. usually #. Advanced users can then use different IVRs to create a multi-tenant installation. These are your options: Change Name: This is simply the descriptive name that appears on the right. in addition to being able to dial the IVR options. be able to directly dial an Extension number. or. 'i'. This may be one.freepbx. 'i' and 't' have special meanings: http://www. users will. Enable Direct Dial: If you enable that. Directory Context: This is the asterisk context of the directory. Configuring your IVR In the box on the left.Administration Guide Page 18 of 42 Unlike the old Digital Receptionist system.You'll see it appear on the right straight away. from the IVR and access the Directory service. This can be set to 'nothing'. or 't'. users will be able to dial the FeatureCodes">feature code for Directory. These announcements are great for “today is July 4th and we're closed for the holiday” and then proceeding on to the regular call 4/20/2011 .

though. and caller pushes 4. A standard configuration is to go the operator. simply leave the selection blank.G. t: This overrides the default timeout behavior. Creating and Assigning Extensions Creating and Assigning Extensions Numbering Schemes There are several schemes for assigning extensions. If you only have 1 2 and 3 defined. click 'Save' and you have your new IVR. To delete an option. When you're finished. queues will not appear as a possible IVR destination if no queues exist. Use 'Increase Options' or 'Decrease Options' to alter the number of options available. Options are only displayed if there is at least one entry created. For example. Invariably. which is to play a 'invalid option' message and immediately replay the current menu. which is to play the menu three times and hangup.Administration Guide Page 19 of 42 i: This overrides the default invalid choice behavior. To test it. give it an incoming route or set up a miscellaneous application (* code) to reach it. 4/20/2011 . This won't let you decrease it to less than the number of options that are currently set.freepbx. you'll find the following http://www. it will jump to this destination. to handle customers that don't have DTMFcapable phones.

.conf applications. choose the last 3 digits of the main number If the main number is 651-3200. For FreePBX. It's usually low enough cost.--------------------------------------------------------------------------------. etc. at minimum. to get the whole block of interest if possible. Don't collide with system shortcuts. Do NOT edit this file as it is auto-generated by FreePBX.Administration Guide Page 20 of 42 guidelines will help: Use their previous extension numbers Upgrading a system shouldn't require upgrading business cards Use the last 3 or 4 digits of their DIDs Less for people to remember For non-DID systems. If the file is owned by FreePBX you should find this statement at the top of the file making it clear that it is owned by FreePBX . 611 and 311 shouldn't be assigned. but the rest of the 600s and 300s can be.conf This file contains the crontab line(s) that will get executed for backup job scheduling.conf alarmreceiver. agents.conf.4.conf backup. 7777 is commonly 'simulate an incoming call'. as should other Miscellaneous Destinations Remember. . or emergency numbers In the US. So here is the list of files as of version 2. then extensions can be 200. and it really hurts to run out. 202. this rules out extensions in the 100s and 900s.conf if you want to use the userfield in the CDR reporting you will need to add this line to the file: userfield=1 then restart Freepbx by typing amportal restart Default file should look like this: http://www. . All modifications to . details at: http://freepbx.--------------------------------------------------------------------------------. should be avoided. Those owned by FreePBX will be in bold underline. 201. custom modifications. There are a few exceptions to this rule but not many. If they become owned in a later version that version will be stated to the right of the file name. when reserving DIDs. .freepbx. There are alternative files to make . cdr_mysql. The basic rule is that all files are owned and modified by FreePBX unless they end _custom. common dialing 4/20/2011 . this file must be done via the web gui.conf asterisk. File ownership and what files you can edit Who owns what files in /etc/asterisk when FreePBX is installed? That's what this page is here to answer. .org/configuration_files .

conf dundi.sock=/tmp/mysql. . This file will not be overwritten. If hostname is not specified .conf if you need to modify existing code code/context in extensions.conf) if so create that context in extensions_custom. If you need to replace the functionality in extensions_additional. port specified or use the default port.if the database server is hosted on the same machine as the .port=3306 . setting hostname=localhost . then cdr_mysql will attempt to connect . you can achieve a local Unix socket connection by .org/book/export/html/1854 4/20/2011 . . to the socket file specified by sock or otherwise use the default socket . extensions_override_freepbx. and additional code enhancements to the FreePBX dial plan. port and sock are both optional parameters.conf enum.conf) has a context or macro that you NEED to modify. Note . This file will not be overwritten. Be very careful as replacing an existing piece of code this way is the fastest possible way to break your system. . you place that code here as asterisk will only execute the first occurrences of that code and ignores other occurrences.freepbx.conf dnsmgr.conf DO NOT EDIT THIS FILE.conf this is the file that you place all your custom contexts.Administration Guide Page 21 of 42 . asterisk server.conf extensions.conf please place your modifications in extensions_override_freepbx. or if hostname is "localhost". If hostname is specified . [global] hostname=localhost dbname=asteriskcdrdb password=amp109 user=asteriskuser . If you need to expand on functionality of a section of code check to see if there is a include context line in the code (will end in _custom. extensions_additional. it get's regenerated each and every time you apply changes. then cdr_mysql will attempt to connect to the . and is not "localhost". http://www.conf (or extensions_additional.conf but read the notes about this file first.conf and it will get called.sock codecs. extensions_custom.conf If extensions.conf please place it in extensions_override_freepbx. file.conf as asterisk uses the code for the first context referance and ignores additional occurances. If you are doing this you should probably think about filing for a feature request or bug fix to get it addressed properly.conf extconfig.

conf parking_additional.conf meetme. queues_additional.conf Do not edit this file in any (should no longer be used as parking was moved to features) phone.conf features_featuremap_custom.conf iax_additional.conf features_applicationmap_additional.conf manager_custom.conf globals_custom.conf features_general_additional.conf meetme_additional.comf files where it will not get removed or replaced.conf iax_registrations_custom.conf 4/20/2011 . Anything you can think of putting in this file can be placed into one of the _custom.conf mgcp.conf musiconhold_additional.conf features_applicationmap_custom.conf Do not edit this file in any way.conf phpagi.conf manager_additional.conf queues.conf iax_general_additional.conf iax_custom_post.conf iax_general_custom.conf iax_custom. Anything you can think of putting in this file can be placed into one of the _custom.conf musiconhold.conf privacy.conf modules.conf features_general_custom.conf files where it will not get removed or replaced.conf modem.conf features_featuremap_additional. http://www. queues_custom.conf file for your queues setup.conf logger.conf iax.conf manager.Administration Guide Page 22 of 42 features.conf localprefixes.conf iaxprov.freepbx.conf oss.conf iax_registrations.conf indications.conf This is the proper location for placing any of the context specific options and lines that you might need to add before the processing of the queues_additional.

res_mysql. if so that is ok as long as the lines only exist in one file and not both (or a big debugging mess will occur along with hair loss as you pull it out while tracking it all down). etc.conf rtp. See sip_nat.conf sip. create a context line: [79](+) then on the next line add the item(s) you need to add. If you do edit this file and place something new in it. http://www. If you need to adjust sip jitter or something else it will be sip_general_custom.conf or if it is a legacy system sip_nat.comf files where it will not get removed or replaced. Some of the required lines for nat'ing are externip=. localnet= (you can have more then one occurrence of this line).conf file.conf.conf for more info.conf This is the proper location for placing any of the [general] context option lines that you might need to add to your setup. sip_general_custom. queues_post_custom. If you want to add additional setup parameters for your sip device see sip_custom_post. If you need to override one of these or add a new one please do so in sip_general_custom. The first three are needed to properly setup a box on protected network behind a firewall that is providing nat to a public IP. If you are looking to do nat'ing.conf in the past.freepbx.conf. This is also the place to add those lines needed to enable the nat'ing of SIP when you go through a firewall. If you have a legacy system these lines might have been placed in sip_nat.conf.conf This is the proper location for placing any of the [general] context option lines that you might need to add to your queues setup.conf Do not edit this file in any way.conf Do not edit this file in any 4/20/2011 . nat=. So for example you have a queue 79 that need a additional parameter added. queues_general_additional.comf files where it will not get removed or replaced.conf This is where FreePBX places all of it's general context settings. Anything you can think of putting in this file can be placed into one of the _custom.conf (if it is for the general context) or sip_custom.conf This is the proper location for placing any of the context specific options that you might need to add to the end queues setup. see sip_general_custom. This is the file that allows you to add or remove values to those entries found in the autogenerated queue_additional. and optionally fromdomain=. To remove use (-) instead followed by the line(s) you want removed.Administration Guide Page 23 of 42 queues_custom_general. sip_general_additional.conf. it will get overwritten at some point and next time you restart your system you will suddenly wonder why things stopped working. Anything you can think of putting in this file can be placed into one of the _custom.

The new preferred location is sip_general_custom. sip_additional.conf http://www.0 sip_nat. If you don't do this the phone system will assume that phones on those other subnets are external and thus provide the External IP of the box in the SIP headers instead of the internal IP.1. sip_notify.0/255.0 localnet=192.conf This is the old common location for placing the lines needed to enable the nat'ing of SIP. sip_registrations_custom.1.conf or sip_nat.conf file.255.conf General section registrations that are auto-generated by FreePBX. If you move the lines from this file to sip_general_custom.conf.0/255. To remove use (-) instead followed by the line(s) you want removed. sip_custom.2 on a 192.0 subnet Requires these two lines in the either sip_general_custom.255.255. Create a context line: [1000](+) then on the next line add the item(s) you need to add. sip_custom_post.168.conf file localnet=192.0/255.255.0 network Phones inside the office are on the 192.conf please remove them from this file or you'll experience hair loss as you spend time debugging why things don't work as you expect.conf a custom file just in case there is ever a need to override a general registration that was autogenerated by FreePBX.168. etc.168.1.freepbx.conf..conf This is the first file that is not under the general context.168.conf This is where FreePBX puts all sip extensions.168. So for example you have an extension 1000 that needs an additional parameter added.2. etc.255.255.255.conf This is the file that allows you to add/remove values to those entries found in the auto-generated sip_additional. trunk.Administration Guide Page 24 of 42 configurations with multiple subnets: For those setups with internal networks that have multiple subnets you will need to add a localnet= line for each subnet that the phone system should have direct access to. IT allows you to define contexts that you need before the contexts that are auto-generated by FreePBX in sip_additional.0/255. sip trunks.2. see 4/20/2011 . If you need to add a additional parameter to a extension. This then becomes a routing problem for the phone as it should not be attempting to talk external IP of the internal box (most firewalls can not handle the looping back of IP traffic). Example: Server 192.conf. sip_registrations.255.

inc this file contains the e-mail subject line and message body for any voice mails that are e-mailed.freepbx. so please be careful. The most common change to this file is to create a context called [zonemessages].inc [default] Once you have configured a system with voicemail there will be values after the context [default].inc file at the initial build time.conf voicemail. If you need to edit the mail sending parameters edit the vm_general. [general] #include #include vm_email.Administration Guide Page 25 of 42 #include vm_email.conf This file is both editable by you and by FreePBX. 99% of the world needs to edit two lines in the [zonemessages] eastern = America/New_York|'vm-received' q 'digits/at' IMp central = America/Chicago|'vm-received' q 'digits/at' IMp mountain = America/Denver|'vm-received' q 'digits/at' IMp pacific = America/Tijuana|'vm-received' q 'digits/at' IMp eastern24 = America/New_York|'vm-received' q 'digits/at' R central24 = America/Chicago|'vm-received' q 'digits/at' R mountain24 = America/Denver|'vm-received' q 'digits/at' R pacific24 = America/Tijuana|'vm-received' q 'digits/at' R deutschland = Europe/Berlin | 'vm-received' Q 'digits/at' kM england = Europe/London | 'vm-received' Q 'digits/at' R germany = Europe/Berlin | 'vm-received' Q 'digits/at' kM alberta = Canada/Mountain | 'vm-received' Q 'digits/at' HM madrid = Europe/Paris|'vm-received' Q 'digits/at' R paris = Europe/Paris|'vm-received' Q 'digits/at' R sthlm = Europe/Stockholm|'vm-recieved' Q 'digits/at' R europa = Europe/Berlin|'vm-received' Q 'digits/at' kM italia = Europe/Rome|'vm-received' Q 'digit/at' HMP military = Zulu | 'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' [default] vm_email. This context allows you to create timezones so that when you have extensions in multiple time zones they can date time stamp recorded messages properly for any given extension. If you create this context it should be placed after the second #include line and before the [default] line. If you are looking to customize the e-mail message that get's send out with a voice mail please edit the vm_email. 4/20/2011 . The structure of this file is as follows: [general] #include vm_general. These lines will be generated by FreePBX every time you add/edit/delete a extension.

this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution.conf zapata_additional. Berkeley nor the names of its contributors may be used to endorse or promote http://www. and the quality isn't all that great. zapata.26MB Download. other common lines to edit are: maxmessage= this is the max message limit. to have all of the asterisk sounds re-recorded.conf zapata_custom_chan_default.) the system 4/20/2011 .conf Hardware examples Add child pages to enter hardware examples here. you'll know that volume and timing can be a bit wonky sometimes. etc. maxmsg= limits the total number of messages allowed in a mailbox. The most common change to this file is to edit the servermail= line so that it is from a valid worldly e-mail address or any mail server that has spam and/or spoofing protection will reject the voice mail e-mails. Here are the links to the files: aLaw Sounds (For use in most Countries) uLaw Sounds (For use in the US) GSM Sounds iLBC Sounds g729 Sounds S-Linear Sounds (The Asterisk Native format) ALL FILES above in one archive for easy installation .freepbx. High Quality Sounds For those using the sounds that come with asterisk. and has released them under the BSD-License for all to use. Kristian Kielhofner of astLinux has come to the rescue by paying. Redistributions in binary form must reproduce the above copyright notice. this list of conditions and the following disclaimer. are permitted provided that the following conditions are met: Redistributions of source code must retain the above copyright notice. Neither the name of the University of California.Administration Guide Page 26 of 42 this file contains the e-mail / voice mail configuration parameters. operator= if this is set to yes then when a person is leaving a message they can press 0 for the operator (or dial another extension).conf zapata-auto. Redistribution and use in source and binary forms. due to them only being in GSM format. Make sure and note what call levels (and conferences. with or without modification. out of his own pocket.

enter the following from the command line. OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE.conf for AAH 1. 1. OR PROFITS. EVEN IF ADVISED OF THE POSSIBILITY OF SUCH 4/20/2011 . STRICT LIABILITY. SPECIAL. 3. 4. THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES. To do that. you will need to edit the following files. INCLUDING. rebuild_zaptel (restart after each command) genzaptelconf (see notes re command switch) Next go into the AMP web interface to create a trunk and you will notice that there is already a trunk called ZAP/g0. zapata. EXEMPLARY. THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.conf and modules.freepbx.Administration Guide Page 27 of 42 products derived from this software without specific prior written permission. DATA. INCIDENTAL. it may be necessary to configure it by using the zaptel card auto-config utility so the correct zaptel driver will be set up. To make outbound calls you will need to set an outbound route as well. zaptel. INDIRECT.1 DIGIUM WILDCARD X100P FXO PCI CARD This card allows you to connect a POTS (plain Old Telephone System) line to your Asterisk@Home box (See Notes for Patch information). You need to edit this.x http://www. BUT NOT LIMITED TO. 2. If you have this card installed.conf. OR CONSEQUENTIAL DAMAGES (INCLUDING. If this card is added after Asterisk has been configured. WHETHER IN CONTRACT. PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES. IN NO EVENT SHALL THE REGENTS AND CONTRIBUTORS BE LIABLE FOR ANY DIRECT. Interfacing to a PSTN 9. OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY. Enter the phone number for you pots line in the Caller ID field Enter 1 for Maximum channels Set a dial rule you want for this trunk Select an outbound dial prefix to select this trunk when dialing Set the Zap Identifier to 1 (the default is g0) Once the card is configured. LOSS OF USE. BUT NOT LIMITED TO. you must add a route for Incoming Calls or asterisk will not answer this line Click on Incoming Calls in AMP and set up an incoming route. 5.

conf Under [channels] edit the following lines: [channels] busydetect=yes busycount=6 For my installation to function correctly.x.3 modules.conf Change the loadzone and defaultzone to 'au' # Global data loadzone = au defaultzone = au 9. 9.Administration Guide Page 28 of 42 or modprobe.x) For AAH 1. 9.1. add the line highlighted in Bold below: .Include AMP Configs channel => 1 #include zapata_additional.conf.1.2 zaptel.1.conf for AAH 2. locate the post-install wcfxo entry and edit it to reflect this: post-install wcfxo /sbin/ztcfg opermode=AUSTRALIA For AAH 2.conf (modprobe.x.freepbx. .conf Leave the rest of the file as it is. alias char-major-196 torisa options wcfxo opermode=AUSTRALIA .x. The last 2 files live in the /etc directory – use a text editor to edit them. It is located at the end of the file. I have also changed the following setting to obtain a good compromise on volume/echoing: rxgain=10.0 (you may have to experiment a little with this setting) txgain=8.0 (you may have to experiment a little with this setting) Ensure the following exist in zapata.1 zapata. add this line 4/20/2011 .conf for AAH 2.

When you open the zapata_auto. http://www. reboot your PC and when Asterisk starts. Create a ZAP trunk in AMP for Channel 2 context=from-pstnchannel => 2 < .Note .1 4/20/2011 . If this card is installed after Asterisk has been loaded.this is a trunk. using config edit.Administration Guide Page 29 of 42 install tor2 /sbin/modprobe --ignore-install tor2 && /sbin/ztcfg . this card has 4 module ports that can be loaded with FXS or FXO modules.freepbx. Set them up as per setting up routes in the earlier chapters of this document.conf .this is an extension.-this would have been defined already by the config signaling=fxs_ks . Similarly. this card allows you to connect a POTS (plain Old Telephone System) line to your Asterisk@Home box.-this would have been defined already by the config If in the illustration it shows channel 1 is your Zap extension then add a zap extension for channel 1 in AMP and if it shows your Zap trunk is channel 2 you should create a zap trunk for channel 2 in AMP.2. 9. Create a ZAP extension in AMP for Channel 1 channel => 1 < . you will need to configure it just like the X100P by using the following command on the command line: genzaptelconf 9. look in the zapata-auto. Set up the trunks as trunks and the extensions as extensions in AMP. use AMP to add a route for incoming calls or asterisk will not answer your trunk.conf file and you will see a list of all your channels in your TDM400P. Channel 1 is the top RJ-45 on the back of the TDM400P card.conf Next. Once this is done. Unlike the X100P. to make outbound calls you will need an outbound route. Note .2 DIGIUM TDM400P FXO/FXS CARD Like the Digium Wildcard X100P. it will look something like the illustration below (see the red highlight) zapata-auto. Span 1: WCTDM/0 'Wildcard TDM400P REV E/F Board 1' signaling=fxo_ks .conf file.

conf (AAH 1. options wctdm opermode=AUSTRALIA fxshonormode=1 bootstringer=1 Your modules. The example below is for AAH 4/20/2011 .aumixrc -L >/dev/null 2>&1 || : pre-remove sound-slot-0 /bin/aumix-minimal -f /etc/. 9. you will need to edit the following files. Do not change anything else.2.aumixrc -S >/dev/null 2>&1 || : alias usb-controller usb-uhci alias char-major-196 torisa options wctdm opermode=AUSTRALIA fxshonormode=1 boostringer=1 options torisa base=0xd0000 post-install tor2 /sbin/ztcfg post-install torisa /sbin/ztcfg post-install wcusb /sbin/ztcfg post-install wcfxo /sbin/ztcfg post-install wctdm /sbin/ztcfg post-install ztdynamic /sbin/ztcfg You will only need to add the line in red.conf.2 modules. in AAH Ver. or modprobe. you may also do the following: Locate the line 'install wctdm /sbin/ztcfg-.conf as per the X100P card in the previous section.x) should look like the example below: alias eth0 e100 alias sound-slot-0 es1370 post-install sound-slot-0 /bin/aumix-minimal -f /etc/.conf (modprobe. where you need to add the following line.conf to add the necessary option for usage in Australia.x) You will need to edit the modules. zapata.conf for AAH 2.--ignore-install http://www.conf and zaptel.1.--ignore-install wctdm && /sbin/ztcfg' and edit it to reflect the following: install wctdm opermode=AUSTRALIA fxshonormode=1 boostringer=1 /sbin/ztcfg-.Administration Guide Page 30 of 42 If you have this card installed. Or.2.x.freepbx.

com/index.EL-i686/ include/linux mv 4/20/2011 . you can start rebuilding the support for your ZAP devices or for that matter. ztdummy if you don’t have any ZAP devices.Administration Guide Page 31 of 42 wctdm && /sbin/ztcfg Note: as of Zaptel Drivers 1.php?p=123 Log into your new server as root and issue the following commands: cd /usr/src/kernels/2. Log in as root and type the following command: rebuild_zaptel Then reboot your system: shutdown -r now Now log in as root again and enter the following command: amportal stop genzaptelconf Reboot once again: shutdown -r now http://www. Unfortunately.4.freepbx.source Nerd Vittles http://nerdvittles.3 (Users Suggestions) fxshonormode=1' Also see Appendix E.h Once the file has been retrieved.2.h.old wget http://nerdvittles.3 REBUILDING ZAPTEL DRIVER Every time there is a kernel update with yum (which is the case with Asterisk and CentOS). The following is the fix . ZAP device support needs to be rebuilt using the new kernel. reboot using the following command: shutdown -r now When the reboot completes.6.9-34. this will cause a slight problem as RedHat bug caused the rebuilding process to fail.h spinlock. by selecting opermode=AUSTRALIA the zaptel drivers automatically add the 'boostringer=1 .

and you're done.5a). SIPURA SPA3000 AS A PSTN INTERFACE To those new to the SPA3000. While it is directed mainly at standalone ATA users. in case you ever want to know what the defaults were. as no one single source of information that I've found so far has a config that would actually work for me.2 Change the settings System tab DHCP: No Static IP: something on your local subnet e. I've put the following together.g.35 Secondary DNS: your ISP's secondary DNS address e. Nothing should have changed in your settings.0 Gateway: your router's IP address e.12. except that you have a few extra options that you didn't have before. 192.freepbx.168... 9. 192.160.html page) of your current SPA-3000 configuration. just save the . Now reset SPA-3000 back to factory defaults.12..e..4.160.g. To help them in their endeavours.1.200 NetMask: 4/20/2011 . I'd suggest taking a snapshot (i.4. Take another snapshot for good measure. then upgrade it to the latest version (at the time of writing. Before you change anything..254 Primary DNS: your ISP's primary DNS address e.255. it gives a good insight of the Sipura SPA3000’s capabilities. (See also user Users’ Suggestions) 9. If you're not already running the latest SPA-3000 firmware. I have come across a few people in the various forums wanting to use their Sipura SPA-3000s as FXO front-end to their Asterisk@Home boxes. because I'm only going to list the minimum changes required to keep things simple. 9. just in case you ever need to refer back to your own customisations.1 Log in to SPA3000 Login to your SPA-3000 as admin/advanced. there is a simplified installation and configuration instruction by JMG Technology.g.g. Take another snapshot now too.Administration Guide Page 32 of 42 . it's 3.36 Regional tab http://www.1.1. 203.

2.2.Administration Guide Page 33 of 42 Dial Tone: 400@-19.0/2/0) Ring 1 Cadence: 60(1.2/4.1..168.2/1+2+3.4/2.4/...2.2/1) Ring Back Tone: 400@-19...2.4/..4/2..4/2.4) Ring 3 Cadence: 60 (1.10(.4/.g..4/.4/.10(.4/.org/book/export/html/1854 4/20/2011 . etc. FXS Port Impedance: 220+820||120nF Line 1 tab Proxy: IP address of your Asterisk box e.4) Hook Flash Timer Min: ..4/2.13 Delete all the Vertical Service Activation Codes..4/.10(*/0/1+2) Busy Tone: 425@-10. 200 Password: password for that extension Silence Threshold: medium DTMF Tx Method: INFO Hook Flash Tx Method: INFO Dial Plan: (*xx|000|0011xxxxxxxxxxx. PSTN Line tab (method 1) http://www..) for example (*xx.450@-19.) will work..4/2) CWT8 Cadence: 30(...4/2..g.4/2.2.2. but I like to do a bit of sanity checking..5/3.5/3.4/2..2/1+2+3.4/.4/2.2..4/.|0[23478]xxxxxxxx|09xxxxxx|1100 |122[135]|1222xxxxxxx|12510[12]|12554|1[38]00xxxxxx|13[1-9]xxx |1747xxxxxxx|2xx|393xxxxxx|3xxxx.4/.2/. 192.4. |[4689]xxxxxxx|7777|899060xxxxx..2/.4/.425@-19..07 Hook Flash Timer Max: ..2.2..2.4/..*(..4/.425@-19.234 Register Expires: 60 Display Name: Whatever User ID: Asterisk extension number e.4/2.4/1) Reorder Tone: 425@-10.freepbx.2.|x.

g.freepbx..Administration Guide Page 34 of 42 Proxy: IP address of your Asterisk box 4/20/2011 .234 Register: no Make Call Without Reg: yes Ans Call Without Reg: yes Display Name: No name User ID: PSTN Password: password Silence Supp Enable: no Echo Canc Enable: no Echo Canc Adapt Enable: no Echo Supp Enable: no FAX CED Detect Enable: yes FAX CNG Detect Enable: yes FAX Passthru Codec: G711u FAX Codec Symmetric: no FAX Passthru Method: None DTMF Tx Method: INFO FAX Process NSE: no Dial Plan 1: (S0<:T0298765432>) for example VoIP Caller Default DP: none PSTN Ring Thru Line 1: no PSTN CID For VoIP CID: yes PSTN Answer Delay: 2 PSTN Ring Thru Delay: 3 PSTN Ring Timeout: 4 http://www.168.1. 192.

org/book/export/html/1854 4/20/2011 .375/1+2) FXO Port Impedance: 220+820||120nF On-Hook Speed: 26ms (Australia) (Source reference: Colin Swan) Or alternatively you may want to adopt the second method for the PSTN Line Tab.freepbx.234 Register: no Make Call Without Reg: yes Ans Call Without Reg: yes Display Name: No name User ID: PSTN Password: password Silence Supp Enable: no Echo Canc Enable: no Echo Canc Adapt Enable: no Echo Supp Enable: no FAX CED Detect Enable: yes FAX CNG Detect Enable: yes FAX Passthru Codec: G711u FAX Codec Symmetric: no FAX Passthru Method: None DTMF Tx Method: INFO FAX Process NSE: no http://www.425@-30. PSTN Line tab (method 2) Proxy: IP address of your Asterisk box e.. 192.1 Disconnect Tone: 425@-30.1(.Administration Guide Page 35 of 42 PSTN Hook Flash Len: .g. which I am currently using.1.375/.168.

User 1 tab Default Ring: 3 Default CWT: 8 4/20/2011 .1(.freepbx.425@-30.4. add a SIP trunk.Administration Guide Page 36 of 42 Dial Plan 1: (S0<:s@YourAsteriskIP>) e.375/. Outbound Caller ID: <0298765432> (for example) Maximum Channels: 1 Dial Rules: 0+NXXXXXXXX (for example) 0011+ZXXXXXXXXXX.g. You may also get CLID if your Telco has activated incoming Caller ID on your phone.101:5060>)or try w/o the port designation VoIP Caller Default DP: none PSTN Ring Thru Line 1: no PSTN CID For VoIP CID: yes PSTN Answer Delay: 2 PSTN Ring Thru Delay: 3 PSTN Ring Timeout: 4 PSTN Hook Flash Len: . (S0<:s@192. you will not need to create an Inbound Route for this channel as the call is sent directly to your “s‿ extension as defined in your incoming call setting.375/1+2) FXO Port Impedance: 220+820||120nF On-Hook Speed: 26ms (Australia) Using this alternative method.168. Trunk Name: telstra (for example) Peer Details: canreinvite=no http://www.3 Add SIP Trunk Then in AMP.1 Disconnect Tone: 425@-30.

200) insecure=very nat=no port=5061 for example secret=password type=user username=PSTN Leave "Register String" empty Then add a DID Route of T0298765432 (for example). 4/20/2011 .168.200) insecure=very nat=no port=5061 (for example) qualify=yes secret=password type=peer username=PSTN User Context: telstra-incoming (for example) User Details: canreinvite=no context=from-pstn host=the IP address of your SPA-3000 (for example. (Source reference: Colin Swan) See the alternative configuration that I am currently using for the PSTN Tab in Notes Also see Eliminating echo problems in Appendix E. 192.168. which goes to your chosen Destination.freepbx.Administration Guide Page 37 of 42 context=from-pstn host=the IP address of your SPA-3000 (for example.4 in Sipura SPA-3000 http://www.1.1.

but if you will be happy with a quality that is not quite but close to your existing PSTN calls.00 from Dick Smith . also play very important roles. Linux CLI Commands Entering the Asterisk Console http://www. you might be in luck. if you want the ability to make PSTN calls. it may not cost you anything at all. Some VSPs like Pennytel. VOIP via the Public Internet is very much dependant on a number of factors – available bandwidth not withstanding. Astratel. If you already have a spare computer to dedicate to this task. If you want to restrict all your calls to VOIP only.00. you will be somewhat disappointed. which may include a monitor. your usage habit of the internet and LAN traffic and equipment 4/20/2011 . Your only other initial cost will be the $20.00 or so activation fee to Oztel (or other VSP of your choice).Australia. Spantalk etc will register you for SIP communication for free provided that you do not need to make PSTN calls. then the cost is almost nothing unless you need to buy an audio headset ($15.) for the softphone. All these “Major Expenses" will be recovered when you receive your monthly Telstra or Optus phone bills. amongst others.freepbx. What will the Quality of the phone calls be? If you are expecting the quality to be as good as your existing PSTN calls. and a Windows PC to run the softphone. the cost will be minimal.Administration Guide Page 38 of 42 Is Voip for You? Is VoIP for You? Whether VOIP is for you or not rely on a number of or combination of factors. then you may be able to buy one from your local swap meets for under $200. What is it going to cost? Assuming that you already have a broadband service. If you do not have a spare PC with the above specification. Some economic and quality considerations should be examined. Ensure that the PC has an Ethernet NIC for connecting to your home network. a router.

02 -c 500 -s 270 <host> Intensive Performance Information vmstat 1 Current Wanpipe Version wanrouter version Current system processes ps aux Current Networking Information ifconfig -a Duplexing Diagnostics mii-tool Rsync Usage rsync -av -essh /path/to/file <remote_site>:/path/to/file SCP Usage scp /path/to/file <remote_host>:/path/to/file Checking Disk Space df -h 4/20/2011 .Administration Guide Page 39 of 42 asterisk -r Checking Current System Load top Interrupt Information cat /proc/interrupts RAID Array Information cat /proc/mdstat Checking the Routing table netstat -rn OR route Checking CPU Information cat /proc/interrupts Checking Memory Information cat /proc/meminfo Running tcpdump tcpdump -A -s 10000 port <port> and host <host> Running PING tests ping -i 0.

org/book/export/html/1854 4/20/2011 . the following are recommended: 1.Administration Guide Page 40 of 42 Setting up Phones Some docs on how to setup up hard and soft phones with freepbx Setting Up Voicemail Setting up Voicemail Voicemailboxes are typically created when used for the first time. Voicemail Locator (VMX) Feature This optional feature lets users set up a short menu before voicemail takes the actual if you don't want to pay for a real one. Voicemail pager feature This gives a short description of the message envelope. Set your hostname to be a fully-qualified domain name.freepbx. be sent an email with both general instructions. Use dyndns. and the system will be insecure. 2.' Resetting It's commonplace to have to reset the passwords as people leave the company. Instructing New Users New users should. as matter of policy. System Tools Area for additional system tools for Asterisk and FreePBX http://www. Don't send from dynamic IP addresses. and directions to change it. Voicemail in email feature In order for these emails to pass through spam filters. suitable for emailing to a wireless carrier's email gateway. 2 for my assistant or just leave a message after the tone. Common options are 'press 1 for my cell. Resist the impulse to standardize the default. Most people won't change it. Choose a password that is at least 4 digits. their voicemail initial password. Assigning Voicemail PasswordsYou must enter a voicemail password when creating an extension enabled with voicemail.

simply do the Webmin make it easy to configure application like SMTP 4/20/2011 .putty. It is written and maintained primarily by Simon Tatham and can be downloaded from the following link.260-1.noarch. along with an xterm terminal emulator.0. If that is the case. etc.rpm I have found the above method is straightforward and simple.shtml You may connect to Webmin remotely through your browser using the following address http://<YourAsterisk_IPAddress>:10000. the alternative installation method can be found here: http://www.dl.2601.Administration Guide Page 41 of 42 Putty PuTTY PuTTY is a free implementation of Telnet and SSH for Win32 and Unix platforms. do the following: wget http://superb-east. rpm –Uvh http://superb-east. http://www. 192.dl. system settings.101:10000 To update WebMin Anytime you want to update Webmin.html WEBMIN WEBMIN Webmin in an invaluable web based gui for managing a Linux box. Or be totally lazy like me and do the whole lot in a one liner. Those who want to use Web Admin to maintain the Asterisk System may download Webmin from here or from CLI.noarch.g.sourceforge.168.terrasoftsolutions.noarch.rpm Install it with the following command through CLI: rpm -Uvh webmin-1.2601. editing files. However there are some users who found that following an alternative method is simpler.sourceforge.

net/eng/index.Administration Guide Page 42 of 42 Log on to your Asterisk box (SSH or at the console). 4/20/2011 .php http://www. At the command prompt.freepbx. issue the following command: yum –y install webmin WINSCP WINSCP WinSCP is an open source freeware SFTP client for Windows using SSH. Its main function is safe copying of files between a local and a remote computer. It can be downloaded from the following link. Legacy SCP protocol is also supported.