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  • Administration Guide
  • Adding Extensions
  • Asterisk CLI Commands
  • Connecting 2 or more boxes
  • Creating Administrator Roles
  • Creating an IVR
  • Digital Receptionist or IVR
  • Creating and Assigning Extensions
  • File ownership and what files you can edit
  • Hardware examples
  • High Quality Sounds
  • Interfacing to a PSTN
  • Is Voip for You?
  • Linux CLI Commands
  • Setting up Phones
  • Setting Up Voicemail
  • System Tools
  • Putty

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



Administration Guide

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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



Administration Guide

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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.



Administration Guide

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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.



Administration Guide Page 5 of 42 accountcode: enter an account code for use by a billing module. 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. mailbox: extension@default is the default.org/book/export/html/1854 4/20/2011 . or specific help on a command include context: Include context in other context load: Load a dynamic module by name logger reload: Reopen 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.freepbx. Use after rotating the log files. debug channel: Enable debugging on a channel dont include: Remove a specified include from context help: Display help list.

conf. 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 AGI Commands http://www. the current values of global variables are not written into the new extensions.conf file with an exported version based on the current state of the dialplan.Administration Guide Page 6 of 42 extensions.2. Using "save dialplan" will result in losing any comments in your current extensions.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 A.conf.freepbx. A backup copy of your old extensions.conf is not saved. The initial values of global variables defined in the [globals] category retain their previous initial values.org/book/export/html/1854 4/20/2011 .

freepbx.org/book/export/html/1854 4/20/2011 .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 A.2.3 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.2.4 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.

2.2.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.7.5 SIP Channel commands sip debug: Enable SIP debugging sip no debug: Disable SIP debugging sip reload: Reload sip.freepbx.conf (added after 0.org/book/export/html/1854 4/20/2011 .

freepbx. I solicited some advise from a friend (thanks to Mark Brooker) who told me that my configuration could be made a lot tidier. to set 2 very basic systems together (you can refer to DUNDi for a more complete solution).au) with about 11 extensions and another office in a different location (xyz. but I prefer 2 separate extensions as I have them working).1 METHOD 1 .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. Avoid using extension starting with 8 as it may clash with conferencing. 27. That I did.com.com.au and xyz. I have 2 different locations. For simplicity. Instead of being verbose in my explanation.org/book/export/html/1854 4/20/2011 . I hope this will help those in the same position as I am.au) together and if you are like me.com. Trunk Name Parramatta http://www. The main office is the only box that will have accounts with different VSPs and all external communications are through the main office Asterisk box.with the peer Asterisk boxes as extensions For the purpose of registering the peers to each other. 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 gave a common password xxxyyy to both boxes. System 1 System 2 IAX Trunk Outgoing Dial Rules: XX.au) about 20 km away with 9 extensions. you will probably be spending a good part of 3 hours trying to get them to talk to one another.com. XX. 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. I will just create a few tables outlining what I did. the Main Office (def.

System 1 System 2 Extensions Phone Protocol IAX IAX Extension Number 90000 91000 Extension Password http://www.com.com. It will still work without it unless you use Dynamic IP.au 90000:xxxyyy@def.au (or IP) secret=xxxyyy type=peer username=91000 host=def.au (or IP) secret=xxxyyy type=peer username=90000 User Context Leave blank Leave blank User Details Leave blank Leave blank Register String 80000:xxxyyy@xyz.com.au Note: Registration isn’t really necessary.freepbx.com.Administration Guide Page 10 of 42 MainOffice Peer Details host=xyz.org/book/export/html/1854 4/20/2011 .

as it requires an extension to be created for the peer Asterisk box. If you have Dynamic IP addresses.org/book/export/html/1854 4/20/2011 . If you are a part of a Corporate LAN. Note: While this method will provide some rudimentary security (though pretty weak). it will http://www. than you will have no need to worry about DynDns and what not. The above example assumes that both Asterisk boxes have Public Fix IP address. 9|6XXX and 9|XX.(Apart from Local extensions. Instead. for system 1 and system 2 respectively instead of just 6XXX and XX. you will need to register both the boxes with DynDns to obtain a valid DNS ID.freepbx. it will not pass the calling party extension number to the remote Asterisk box. you may place a prefix e.g. 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.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. If you want to use a prefix to dial the remote extensions and to use the remote routing rules.

Trunk Name InterOffice InterOffice Peer Details host=xyz. Like all installation. System 1 System 2 IAX2 Trunk Outgoing Dial Rules: 6XXX XX. however I believe. This method does not require registration either and does not require you to create extensions for the peers. The receiving party will actually get the callers’ extension number/ID instead of the extension number of the peer Asterisk box.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.freepbx.com. Rather than being verbose. I will illustrate this method using tables as follows.org/book/export/html/1854 4/20/2011 . I will leave that to the individual implementer to deal with the security issues. This method treats both the Asterisk box as internal to each other as peer and user. I am using IAX2 for this purpose. this is simpler to set up. 27. you must provide for security. In many ways.au (or IP) Qualify=yes http://www. As different installation resorts to different types of security arrangement. Note: You must provide for security. (Note: A little tutorial on DUNDi can be found here). Unlike the first method.2 METHOD 2 . this second method will pass the Caller ID to the receiving party. 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. as this is pretty wide open.

all others go via City Office) Trunk Sequence IAX2/InterOffice IAX2/InterOffice Thinking of more than 2 boxes? http://www.freepbx.au (or IP) type=user context=from-internal host=def.com.com.au (or IP) Qualify=yes type=peer User Context InterOffice-In InterOffice-In User Details context=from-internal host=xyz.au (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.(Apart from Local extensions.org/book/export/html/1854 4/20/2011 .Administration Guide Page 13 of 42 type=peer host=def.com.

while useable for a basic configuration. Creating Administrator Roles Creating Administrator Roles For most web applications it is useful to have graduated permission access. without exposing trunks and other settings they do not need.freepbx. B and C (System 1. all external and inter-office (inter-branch) traffic goes via Box A. A good policy is to only allow local (LAN) or tunneled via SSH access to the web application. In addition to the webapp username / password settings. se same principle can be applied to more boxes. Both the above methods. though exceptions can be be made for the Recordings (ARI) interface. Mie is allowed to see status.org/wiki/view/Asterisk+dual+servers If you require a complete solution tailored to your exact requirement. – with the appropriate dial plan of course. for example. A peers with B and C .Administration Guide Page 14 of 42 Just as a matter of interest. my advise to you is to hire a VOIP consultant. In my implementation I have box A. both Apache and iptables can be used to restrict access on a location basis to the web application. Box A is the master box. The following link will provide further reference for connecting two Asterisk boxes together http://www. To provide a complete solution is beyond the scope of this document. Show below is just such a configuration. 2 and 3).voip-info. http://www. access to the Extensions directory to change usernames and reset voicemail passwords as employees come and go.And C peers with A. so that users have only access to the functions they need. edit extensions (this part is not shown) and apply changes. This lets you give office managers. Except for local traffic.B peers with A . All the other boxes use box A as the main exchange.I believe. While I have connected 3 boxes successfully.org/book/export/html/1854 4/20/2011 . In this case. you can connect several boxes using this method. will not provide you with a complete solution.

Run it by the customer (or your officemates).freepbx. One way to do this is use miscellaneous destinations. you should resist this impulse. press 5 for office directions. and then make the inevitable changes. 7. ring groups. etc. If you know the room # of the guest you are trying to reach. you may dial it at any time. for Sales press 1. Test all of these. press 3 for administration. If you know the extension of the person you are trying to reach.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 PBX. Please listen carefully as our options have changed. press 2 for customer service. Create any destinations that don't currently exist (queues. Office / Light industrial 1.) 9. Planning While the urge is strong just to dive in by clicking on IVR. day/night modes or time conditions). etc. 5. Now upgrade the voice prompts to a paid voice or designated employee (the office manager or receptionist. assigning a * feature code to whatever thing you want to test. draw out on paper what you intend to to achieve. Bask in glory! Standard IVR Examples: 1. Planning 2. Please listen carefully as our options have changed. 6. 4. 3.org/book/export/html/1854 4/20/2011 . Press 1 for sales. press 4 for Press inquiries.press # to access the company directory. Welcome to BUSINESSNAME. First. 8. you may dial it at any time. Welcome to HOTELNAME. for Service press 2". Record the audio prompts using System Recordings and an extension. Write out word-for-word what all the recordings are going to be. Normally heard as "Thanks you for calling MYBUSINESS. Customer agreement with the plan. Press 1 http://www. Hospitality 1. Then go create your IVR. 2. Show it to the customer. The proper flow to build a good IVR is: 1. or press 0 for the operator.

Welcome to BUSINESSNAME. you may dial it at any time. Shown here is 3. I strongly suggest you use an extension connected to the PBX to make your recordings. you may dial it at any time. Engineering/Product Company with Direct Sales and Support 1.5. press # to access the company directory.3. press 3 for technical support. If you know the extension of the person you are trying to reach. or press 0 for the operator. locations.org/book/export/html/1854 4/20/2011 . 4. Please listen carefully as our options have changed.freepbx. press 6 for office directions. press 5 for hotel directions. They'll be quick http://www. 3. press 5 for Press inquiries. Retail 1. press 4 for administration. If you know the extension of the person you are trying to reach. Making recordings Fire up the System Recordings module. press 2 for customer service. press 2 for customer service.Administration Guide Page 16 of 42 for reservations.1. and directions. Press 1 for sales. Press 1 for sales. press 3 for event sales. or press 0 for the operator. Welcome to BUSINESSNAME. press 2 for the front desk. press 3 for store hours. press 5 for Press inquiries. press # to access the company directory. press 4 for hotel administration. press 4 for administration. Please listen carefully as our options have changed. press # to access the hotel directory. or press 0 for the operator.

you can come back and replace those temporary recordings with paid or improved versions. we can create our IVR. If the recording is good enough (and don't obsess here yet). the first page is now a brief set of instructions on how to drive the IVR. or create a new one by clicking on 'Add IVR'. so the single Welcome-to-ACME recording will be enough. name the recording and press Save.Administration Guide Page 17 of 42 and in the right format and you can worry about getting everything else right. When everything is all finished. Editing your IVR http://www. or you'll get a cryptic error. enter your extension in Step 1 and press Go. Now dial *77 and make your recording after the beep. For lame and silly reasons. if one is existing. 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. Now that we've created a system recording. Dial *99 to listen to it.freepbx. Creating the IVR When you select IVR.org/book/export/html/1854 4/20/2011 . spaces are not allowed in the names. You can either edit an IVR. 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. To use your extension to make a recording. Don't skip this and go to Step 2. We're going to start with a simple 1-level IVR .

or 't'. users will be able to dial the FeatureCodes">feature code for Directory. This can be set to 'nothing'. from the IVR and access the Directory service. These are your options: Change Name: This is simply the descriptive name that appears on the right. or a series of numbers.freepbx. Announcement: A System Recording that is played to users when they enter the IVR. 'i' and 't' have special meanings: http://www. This may be one.org/book/export/html/1854 4/20/2011 . Configuring your IVR In the box on the left. enter the option for the user. this creates the IVR (and calls it 'Unnamed') as soon as you click 'Add' .Administration Guide Page 18 of 42 Unlike the old Digital Receptionist system. in addition to being able to dial the IVR options. or. be able to directly dial an Extension number. Directory Context: This is the asterisk context of the directory. Advanced users can then use different IVRs to create a multi-tenant installation. Enable Direct Dial: If you enable that. These announcements are great for “today is July 4th and we're closed for the holiday” and then proceeding on to the regular call flow.You'll see it appear on the right straight away. users will. usually #. 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. 'i'.

to handle customers that don't have DTMFcapable phones. Use 'Increase Options' or 'Decrease Options' to alter the number of options available. queues will not appear as a possible IVR destination if no queues exist. it will jump to this destination. This won't let you decrease it to less than the number of options that are currently set. When you're finished. To test it.Administration Guide Page 19 of 42 i: This overrides the default invalid choice behavior. simply leave the selection blank. If you only have 1 2 and 3 defined. Options are only displayed if there is at least one entry created. A standard configuration is to go the operator. give it an incoming route or set up a miscellaneous application (* code) to reach it. which is to play the menu three times and hangup. you'll find the following http://www.org/book/export/html/1854 4/20/2011 .G. though. click 'Save' and you have your new IVR. E.freepbx. Invariably. To delete an option. For example. Creating and Assigning Extensions Creating and Assigning Extensions Numbering Schemes There are several schemes for assigning extensions. and caller pushes 4. t: This overrides the default timeout behavior. which is to play a 'invalid option' message and immediately replay the current menu.

. details at: http://freepbx.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.conf This file contains the crontab line(s) that will get executed for backup job scheduling. 7777 is commonly 'simulate an incoming call'.--------------------------------------------------------------------------------. 611 and 311 shouldn't be assigned. 202.freepbx. this rules out extensions in the 100s and 900s.4.conf backup. For FreePBX. So here is the list of files as of version 2. . . to get the whole block of interest if possible. common dialing sequences. Don't collide with system shortcuts. this file must be done via the web gui.org/configuration_files .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. There are alternative files to make . as should other Miscellaneous Destinations Remember. at minimum. agents. and it really hurts to run out. It's usually low enough cost. custom modifications. cdr_mysql. If they become owned in a later version that version will be stated to the right of the file name. 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.conf asterisk. when reserving DIDs. etc. . 201. but the rest of the 600s and 300s can be.--------------------------------------------------------------------------------. Those owned by FreePBX will be in bold underline. There are a few exceptions to this rule but not many. . The basic rule is that all files are owned and modified by FreePBX unless they end _custom. Do NOT edit this file as it is auto-generated by FreePBX.conf. then extensions can be 200. should be avoided.conf alarmreceiver. All modifications to . choose the last 3 digits of the main number If the main number is 651-3200.conf applications. 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 .org/book/export/html/1854 4/20/2011 . or emergency numbers In the US.

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

conf localprefixes. http://www.conf globals_custom.org/book/export/html/1854 4/20/2011 .conf file for your queues setup.conf iaxprov. Anything you can think of putting in this file can be placed into one of the _custom.conf oss.inc (should no longer be used as parking was moved to features) phone.comf files where it will not get removed or replaced.Administration Guide Page 22 of 42 features.conf manager_additional.conf indications. queues_custom.conf iax_registrations.conf iax_custom_post.conf iax_additional.conf Do not edit this file in any way.conf modules.conf meetme_additional.conf features_applicationmap_custom.conf features_general_custom.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.conf phpagi.conf iax_custom.conf files where it will not get removed or replaced.conf meetme.conf features_general_additional.freepbx.conf iax_registrations_custom.conf musiconhold.conf queues.conf iax_general_custom.conf iax_general_additional. queues_additional.conf privacy.conf mgcp.conf iax.conf manager_custom.conf features_applicationmap_additional.conf parking_additional.conf features_featuremap_custom.conf Do not edit this file in any way.conf musiconhold_additional.conf musiconhold_custom.conf manager.conf modem.conf features_featuremap_additional. Anything you can think of putting in this file can be placed into one of the _custom.conf logger.

conf This is where FreePBX places all of it's general context settings. 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).conf file.conf sip.conf. 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.conf. queues_general_additional. it will get overwritten at some point and next time you restart your system you will suddenly wonder why things stopped working. So for example you have a queue 79 that need a additional parameter added.conf Do not edit this file in any way. sip_general_additional. This is the file that allows you to add or remove values to those entries found in the autogenerated queue_additional.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. Anything you can think of putting in this file can be placed into one of the _custom. If you are looking to do nat'ing. nat=. If you have a legacy system these lines might have been placed in sip_nat. To remove use (-) instead followed by the line(s) you want removed. localnet= (you can have more then one occurrence of this line).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. This is also the place to add those lines needed to enable the nat'ing of SIP when you go through a firewall.Administration Guide Page 23 of 42 queues_custom_general. http://www.conf. sip_general_custom.conf Do not edit this file in any way. The first three are needed to properly setup a box on protected network behind a firewall that is providing nat to a public IP. and optionally fromdomain=.conf. etc.freepbx. create a context line: [79](+) then on the next line add the item(s) you need to add. see sip_general_custom. queues_post_custom. If you want to add additional setup parameters for your sip device see sip_custom_post.conf This is the proper location for placing any of the [general] context option lines that you might need to add to your setup.conf in the past.comf files where it will not get removed or replaced. res_mysql. If you need to override one of these or add a new one please do so in sip_general_custom.comf files where it will not get removed or replaced.conf rtp.conf (if it is for the general context) or sip_custom.conf for more info. If you do edit this file and place something new in it. Anything you can think of putting in this file can be placed into one of the _custom.org/book/export/html/1854 4/20/2011 . See sip_nat. Some of the required lines for nat'ing are externip=.

0 sip_nat. The new preferred location is sip_general_custom. If you move the lines from this file to sip_general_custom.conf This is the first file that is not under the general context.168.0 localnet=192. sip_registrations_custom. sip_registrations.168.255.conf or sip_nat.org/book/export/html/1854 4/20/2011 .conf This is where FreePBX puts all sip extensions.conf General section registrations that are auto-generated by FreePBX.conf http://www.1.0 network Phones inside the office are on the 192.255. Example: Server sip trunks. 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.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.0/255.1. trunk.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. If you need to add a additional parameter to a extension.0 subnet Requires these two lines in the either sip_general_custom. sip_additional. see sip_custom_post.conf.conf. IT allows you to define contexts that you need before the contexts that are auto-generated by FreePBX in sip_additional.168.2 on a 192. sip_custom_post.freepbx. sip_custom.255. Create a context line: [1000](+) then on the next line add the item(s) you need to add.255.conf a custom file just in case there is ever a need to override a general registration that was autogenerated by FreePBX. sip_notify.1. To remove use (-) instead followed by the line(s) you want removed. etc. 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).168.conf file.255.0/255.255.0/255. etc..conf This is the old common location for placing the lines needed to enable the nat'ing of SIP.conf file localnet=192.255. So for example you have an extension 1000 that needs an additional parameter added.0/255.conf This is the file that allows you to add/remove values to those entries found in the auto-generated sip_additional.

inc [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.inc #include vm_email. [general] #include vm_general. The most common change to this file is to create a context called [zonemessages].conf voicemail. The structure of this file is as follows: [general] #include vm_general. vm_general.freepbx. If you are looking to customize the e-mail message that get's send out with a voice mail please edit the vm_email.conf This file is both editable by you and by FreePBX.inc file.Administration Guide Page 25 of 42 skinny.inc #include vm_email. If you create this context it should be placed after the second #include line and before the [default] line.inc file.inc http://www.org/book/export/html/1854 4/20/2011 . so please be careful.inc [default] Once you have configured a system with voicemail there will be values after the context [default].inc this file contains the e-mail subject line and message body for any voice mails that are e-mailed. If you need to edit the mail sending parameters edit the vm_general. 99% of the world needs to edit two lines in the vm_general. These lines will be generated by FreePBX every time you add/edit/delete a extension.inc file at the initial build time. 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.

org/book/export/html/1854 4/20/2011 . etc. Berkeley nor the names of its contributors may be used to endorse or promote http://www. due to them only being in GSM format. you'll know that volume and timing can be a bit wonky sometimes. Neither the name of the University of California. Make sure and note what call levels (and conferences. out of his own pocket.) the system acheives. 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 . and has released them under the BSD-License for all to use. with or without modification.conf zapata_custom_chan_default.26MB Download. Kristian Kielhofner of astLinux has come to the rescue by paying.conf zapata_additional.freepbx. zapata. Redistributions in binary form must reproduce the above copyright notice.conf zapata-auto.conf Hardware examples Add child pages to enter hardware examples here. are permitted provided that the following conditions are met: Redistributions of source code must retain the above copyright notice. 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). maxmsg= limits the total number of messages allowed in a mailbox. other common lines to edit are: maxmessage= this is the max message limit. 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. this list of conditions and the following disclaimer.Administration Guide Page 26 of 42 this file contains the e-mail / voice mail configuration parameters. to have all of the asterisk sounds re-recorded. and the quality isn't all that great. this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. Redistribution and use in source and binary forms. High Quality Sounds For those using the sounds that come with asterisk.

4.freepbx. you will need to edit the following files.conf and modules. it may be necessary to configure it by using the zaptel card auto-config utility so the correct zaptel driver will be set up. 5. You need to edit this. zaptel.conf for AAH 1. IN NO EVENT SHALL THE REGENTS AND CONTRIBUTORS BE LIABLE FOR ANY DIRECT. OR CONSEQUENTIAL DAMAGES (INCLUDING. INCLUDING. If you have this card installed. 2. 1.x http://www. THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES. enter the following from the command line. To make outbound calls you will need to set an outbound route as well. 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. If this card is added after Asterisk has been configured. WHETHER IN CONTRACT. LOSS OF USE. BUT NOT LIMITED TO. zapata. OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY.conf.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). 3. OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE.Administration Guide Page 27 of 42 products derived from this software without specific prior written permission. OR PROFITS. SPECIAL. PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES. EXEMPLARY. To do that. EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. INCIDENTAL. 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. Interfacing to a PSTN 9. STRICT 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. BUT NOT LIMITED TO. INDIRECT. DATA.org/book/export/html/1854 4/20/2011 .

conf for AAH 2. I have also changed the following setting to obtain a good compromise on volume/echoing: rxgain=10. add the line highlighted in Bold below: . add this line http://www. The last 2 files live in the /etc directory – use a text editor to edit them.x.conf Under [channels] edit the following lines: [channels] busydetect=yes busycount=6 For my installation to function correctly.conf.conf for AAH 2.org/book/export/html/1854 4/20/2011 .conf (modprobe. 9.1. 9.conf Change the loadzone and defaultzone to 'au' # Global data loadzone = au defaultzone = au 9. locate the post-install wcfxo entry and edit it to reflect this: post-install wcfxo /sbin/ztcfg opermode=AUSTRALIA For AAH 2.Include AMP Configs channel => 1 #include zapata_additional.x) For AAH 1.x.0 (you may have to experiment a little with this setting) Ensure the following exist in zapata.2 zaptel.1 zapata.1.Administration Guide Page 28 of 42 or modprobe.3 modules.x.freepbx. It is located at the end of the file. alias char-major-196 torisa options wcfxo opermode=AUSTRALIA .conf Leave the rest of the file as it is. .1.0 (you may have to experiment a little with this setting) txgain=8.

this card has 4 module ports that can be loaded with FXS or FXO modules. Channel 1 is the top RJ-45 on the back of the TDM400P card. Set them up as per setting up routes in the earlier chapters of this document.org/book/export/html/1854 4/20/2011 . http://www.conf Next. If this card is installed after Asterisk has been loaded. Once this is done.-this would have been defined already by the config signaling=fxs_ks .this is a trunk. 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. Similarly. When you open the zapata_auto. 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. reboot your PC and when Asterisk starts. Create a ZAP trunk in AMP for Channel 2 context=from-pstnchannel => 2 < . Note .Note .-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 DIGIUM TDM400P FXO/FXS CARD Like the Digium Wildcard X100P.freepbx. Set up the trunks as trunks and the extensions as extensions in AMP.conf . to make outbound calls you will need an outbound route.Administration Guide Page 29 of 42 install tor2 /sbin/modprobe --ignore-install tor2 && /sbin/ztcfg .this is an extension. Create a ZAP extension in AMP for Channel 1 channel => 1 < .2. using config edit. Span 1: WCTDM/0 'Wildcard TDM400P REV E/F Board 1' signaling=fxo_ks . this card allows you to connect a POTS (plain Old Telephone System) line to your Asterisk@Home box. 9.1 zapata-auto. it will look something like the illustration below (see the red highlight) zapata-auto.conf file. Unlike the X100P.

9.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.Administration Guide Page 30 of 42 If you have this card installed. or modprobe.org/book/export/html/1854 4/20/2011 . you may also do the following: Locate the line 'install wctdm /sbin/ztcfg-.x) You will need to edit the modules.conf to add the necessary option for usage in Australia.conf for AAH 2.2.conf and zaptel. Or.1.aumixrc -L >/dev/null 2>&1 || : pre-remove sound-slot-0 /bin/aumix-minimal -f /etc/. in AAH Ver.--ignore-install wctdm && /sbin/ztcfg' and edit it to reflect the following: install wctdm opermode=AUSTRALIA fxshonormode=1 boostringer=1 /sbin/ztcfg-.--ignore-install http://www.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/.2.freepbx.x. you will need to edit the following files.2 modules. where you need to add the following line. zapata.conf as per the X100P card in the previous section.conf. The example below is for AAH 1. Do not change anything else.conf (modprobe.conf (AAH 1. options wctdm opermode=AUSTRALIA fxshonormode=1 bootstringer=1 Your modules.

com/aah27/spinlock. by selecting opermode=AUSTRALIA the zaptel drivers automatically add the 'boostringer=1 .6.com/index. Unfortunately. ZAP device support needs to be rebuilt using the new kernel.3 REBUILDING ZAPTEL DRIVER Every time there is a kernel update with yum (which is the case with Asterisk and CentOS).php?p=123 Log into your new server as root and issue the following commands: cd /usr/src/kernels/2. you can start rebuilding the support for your ZAP devices or for that matter.3 (Users Suggestions) 9.source Nerd Vittles http://nerdvittles. reboot using the following command: shutdown -r now When the reboot completes. 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.freepbx.EL-i686/ include/linux mv spinlock.9-34.h.4. fxshonormode=1' Also see Appendix E. ztdummy if you don’t have any ZAP devices.Administration Guide Page 31 of 42 wctdm && /sbin/ztcfg Note: as of Zaptel Drivers 1.h Once the file has been retrieved.2.h spinlock.old wget http://nerdvittles. this will cause a slight problem as RedHat bug caused the rebuilding process to fail.org/book/export/html/1854 4/20/2011 . The following is the fix .

While it is directed mainly at standalone ATA users.org/book/export/html/1854 4/20/2011 . it's 3.200 NetMask: 255.4 SIPURA SPA3000 AS A PSTN INTERFACE To those new to the SPA3000.g.4..e.html page) of your current SPA-3000 configuration.2 Change the settings System tab DHCP: No Static IP: something on your local subnet e. Take another snapshot now too.254 Primary DNS: your ISP's primary DNS address e. 9. 9. Take another snapshot for good measure.168. To help them in their endeavours. I've put the following together.1.Administration Guide Page 32 of 42 . there is a simplified installation and configuration instruction by JMG Technology. 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.12.160. as no one single source of information that I've found so far has a config that would actually work for me.freepbx.g.12.168.g. Nothing should have changed in your settings. Before you change anything.255.1.g.4..36 Regional tab http://www... (See also user Users’ Suggestions) 9. If you're not already running the latest SPA-3000 firmware. just in case you ever need to refer back to your own customisations. in case you ever want to know what the defaults were.1.0 Gateway: your router's IP address e. because I'm only going to list the minimum changes required to keep things simple. just save the . I'd suggest taking a snapshot (i..35 Secondary DNS: your ISP's secondary DNS address e. 203. except that you have a few extra options that you didn't have before. 192. 192. Now reset SPA-3000 back to factory defaults. 203.160.1 Log in to SPA3000 Login to your SPA-3000 as admin/advanced. then upgrade it to the latest version (at the time of writing.255.5a).and you're done. it gives a good insight of the Sipura SPA3000’s capabilities.

.4) Ring 3 Cadence: 60 (1.4/2.0/2/0) Ring 1 Cadence: 60(1.4/2.....1.10(*/0/1+2) Busy Tone: 425@-10.4/. 192.4/..2/1) Ring Back Tone: 400@-19.2..4/1) Reorder Tone: 425@-10. PSTN Line tab (method 1) http://www.234 Register Expires: 60 Display Name: Whatever User ID: Asterisk extension number e.425@-19.4/.2/1+2+3.4/2.4) Hook Flash Timer Min: . FXS Port Impedance: 220+820||120nF Line 1 tab Proxy: IP address of your Asterisk box e...2.4/...4/.4/.|0[23478]xxxxxxxx|09xxxxxx|1100 |122[135]|1222xxxxxxx|12510[12]|12554|1[38]00xxxxxx|13[1-9]xxx |1747xxxxxxx|2xx|393xxxxxx|3xxxx.4.4/.g.4/. but I like to do a bit of sanity checking.2.) for example (*xx.13 Delete all the Vertical Service Activation Codes..2/1+2+3..4/2.) will work.Administration Guide Page 33 of 42 Dial Tone: 400@-19.g.450@-19.4/2) CWT8 Cadence: 30(.4/.4/.5/3.4/.4/.4/2..freepbx...2/..2/4.2.org/book/export/html/1854 4/20/2011 .10(.10(.2.4/.4/2.5/3..4/2.07 Hook Flash Timer Max: ..2/..2... 200 Password: password for that extension Silence Threshold: medium DTMF Tx Method: INFO Hook Flash Tx Method: INFO Dial Plan: (*xx|000|0011xxxxxxxxxxx..2.*(.4/2.|x.2. |[4689]xxxxxxx|7777|899060xxxxx.2..425@-19.2.168. etc.4/2.2.

freepbx.g.168.org/book/export/html/1854 4/20/2011 . 192.Administration Guide Page 34 of 42 Proxy: IP address of your Asterisk box e.1.234 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..

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.Administration Guide Page 35 of 42 PSTN Hook Flash Len: .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.. 192.375/. which I am currently using.425@-30.1 Disconnect Tone: 425@-30.g.1.org/book/export/html/1854 4/20/2011 .1(.freepbx. PSTN Line tab (method 2) Proxy: IP address of your Asterisk box e.168.

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: . add a SIP trunk.1 Disconnect Tone: 425@-30.1(.org/book/export/html/1854 4/20/2011 . Outbound Caller ID: <0298765432> (for example) Maximum Channels: 1 Dial Rules: 0+NXXXXXXXX (for example) 0011+ZXXXXXXXXXX.425@-30. User 1 tab Default Ring: 3 Default CWT: 8 9.168.g.3 Add SIP Trunk Then in AMP.0.Administration Guide Page 36 of 42 Dial Plan 1: (S0<:s@YourAsteriskIP>) e. You may also get CLID if your Telco has activated incoming Caller ID on your phone.375/1+2) FXO Port Impedance: 220+820||120nF On-Hook Speed: 26ms (Australia) Using this alternative method.375/.4. 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. Trunk Name: telstra (for example) Peer Details: canreinvite=no http://www. (S0<:s@192.freepbx.

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. which goes to your chosen Destination. (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.org/book/export/html/1854 4/20/2011 .4 in Sipura SPA-3000 http://www. 192.168.200) 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). Guide Page 37 of 42 context=from-pstn host=the IP address of your SPA-3000 (for example.freepbx.

you will be somewhat disappointed. then the cost is almost nothing unless you need to buy an audio headset ($15. and a Windows PC to run the softphone.00 or so activation fee to Oztel (or other VSP of your choice). your usage habit of the internet and LAN traffic and equipment quality. amongst others. then you may be able to buy one from your local swap meets for under $200. if you want the ability to make PSTN calls. Some economic and quality considerations should be examined.00. the cost will be minimal. Some VSPs like Pennytel.) for the softphone.freepbx. Linux CLI Commands Entering the Asterisk Console http://www. Spantalk etc will register you for SIP communication for free provided that you do not need to make PSTN calls. If you already have a spare computer to dedicate to this task. which may include a monitor.00 from Dick Smith . Ensure that the PC has an Ethernet NIC for connecting to your home network. but if you will be happy with a quality that is not quite but close to your existing PSTN calls. VOIP via the Public Internet is very much dependant on a number of factors – available bandwidth not withstanding.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. you might be in luck. What is it going to cost? Assuming that you already have a broadband service. a router. What will the Quality of the phone calls be? If you are expecting the quality to be as good as your existing PSTN calls.Australia. All these “Major Expenses" will be recovered when you receive your monthly Telstra or Optus phone bills. also play very important roles. Astratel. If you do not have a spare PC with the above specification. it may not cost you anything at all.org/book/export/html/1854 4/20/2011 . If you want to restrict all your calls to VOIP only. Your only other initial cost will be the $20.

freepbx.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 0.02 -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 http://www.org/book/export/html/1854 4/20/2011 .

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

noarch.dl.com/support/solutions/ydl_general/webmin.html WEBMIN WEBMIN Webmin in an invaluable web based gui for managing a Linux box.Administration Guide Page 41 of 42 Putty PuTTY PuTTY is a free implementation of Telnet and SSH for Win32 and Unix platforms.terrasoftsolutions.noarch.putty.2601.sourceforge. do the following: wget http://superb-east.shtml You may connect to Webmin remotely through your browser using the following address http://<YourAsterisk_IPAddress>:10000. Webmin make it easy to configure application like SMTP mail.0. rpm –Uvh http://superb-east. It is written and maintained primarily by Simon Tatham and can be downloaded from the following link. However there are some users who found that following an alternative method is simpler. E. along with an xterm terminal emulator.sourceforge.rpm Install it with the following command through CLI: rpm -Uvh webmin-1.nl/download. http://www.101:10000 To update WebMin Anytime you want to update Webmin.g.260-1.rpm I have found the above method is straightforward and simple.dl. If that is the case.org/book/export/html/1854 4/20/2011 . etc.freepbx. the alternative installation method can be found here: http://www. Those who want to use Web Admin to maintain the Asterisk System may download Webmin from here or from CLI. system settings. http://www.rpm Or be totally lazy like me and do the whole lot in a one liner.net/sourceforge/webadmin/webmin-1. simply do the following.noarch.2601.168.net/sourceforge/webadmin/webmin-1. editing files. 192.

issue the following command: yum –y install webmin WINSCP WINSCP WinSCP is an open source freeware SFTP client for Windows using SSH. It can be downloaded from the following link.net/eng/index.freepbx.Administration Guide Page 42 of 42 Log on to your Asterisk box (SSH or at the console).org/book/export/html/1854 4/20/2011 . Legacy SCP protocol is also supported. At the command prompt.php http://www. Its main function is safe copying of files between a local and a remote computer. http://winscp.

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