freepbx

March 31, 2018 | Author: jankrum | Category: Session Initiation Protocol, Voice Over Ip, Interactive Voice Response, Ip Address, Internet Protocols
Share
Report this link


Comments



Description

Administration GuidePage 1 of 42 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 http://www.freepbx.org/book/export/html/1854 4/20/2011 Administration Guide Page 2 of 42 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 http://www.freepbx.org/book/export/html/1854 4/20/2011 Administration Guide Page 3 of 42 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. http://www.freepbx.org/book/export/html/1854 4/20/2011 Administration Guide Page 4 of 42 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. http://www.freepbx.org/book/export/html/1854 4/20/2011 org/book/export/html/1854 4/20/2011 .freepbx. 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. 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. mailbox: extension@default is the default. debug channel: Enable debugging on a channel dont include: Remove a specified include from context help: Display help list. 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.Administration Guide Page 5 of 42 accountcode: enter an account code for use by a billing module. conf.conf file with an exported version based on the current state of the dialplan.org/book/export/html/1854 4/20/2011 .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 is not saved.freepbx.2 AGI Commands http://www. A backup copy of your old extensions. Using "save dialplan" will result in losing any comments in your current extensions.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: . The initial values of global variables defined in the [globals] category retain their previous initial values.2. the current values of global variables are not written into the new extensions.Administration Guide Page 6 of 42 extensions. freepbx.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.org/book/export/html/1854 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 A.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. 5 SIP Channel commands sip debug: Enable SIP debugging sip no debug: Disable SIP debugging sip reload: Reload sip.freepbx.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.2.2.conf (added after 0.6 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.org/book/export/html/1854 4/20/2011 .7. 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. For simplicity. That I did. 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. 27. 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.au) about 20 km away with 9 extensions.freepbx. XX.au and xyz. Instead of being verbose in my explanation.com.with the peer Asterisk boxes as extensions For the purpose of registering the peers to each other.com.au) together and if you are like me.1 METHOD 1 . System 1 System 2 IAX Trunk Outgoing Dial Rules: XX.com. Avoid using extension starting with 8 as it may clash with conferencing.com.au) with about 11 extensions and another office in a different location (xyz. you will probably be spending a good part of 3 hours trying to get them to talk to one another. 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. I hope this will help those in the same position as I am. I will just create a few tables outlining what I did. I have 2 different locations. I solicited some advise from a friend (thanks to Mark Brooker) who told me that my configuration could be made a lot tidier.org/book/export/html/1854 4/20/2011 . but I prefer 2 separate extensions as I have them working). the Main Office (def. to set 2 very basic systems together (you can refer to DUNDi for a more complete solution). org/book/export/html/1854 4/20/2011 .freepbx.au 90000:[email protected] (or IP) secret=xxxyyy type=peer username=90000 User Context Leave blank Leave blank User Details Leave blank Leave blank Register String 80000:[email protected] Note: Registration isn’t really necessary. It will still work without it unless you use Dynamic IP.au (or IP) secret=xxxyyy type=peer username=91000 host=def.com. System 1 System 2 Extensions Phone Protocol IAX IAX Extension Number 90000 91000 Extension Password http://www.Administration Guide Page 10 of 42 MainOffice Peer Details host=xyz.com. it will not pass the calling party extension number to the remote Asterisk box. If you want to use a prefix to dial the remote extensions and to use the remote routing rules. you will need to register both the boxes with DynDns to obtain a valid DNS ID.freepbx. than you will have no need to worry about DynDns and what not. 9|6XXX and 9|XX. you may place a prefix e.org/book/export/html/1854 4/20/2011 . Instead. it will http://www.(Apart from Local extensions. If you are a part of a Corporate LAN. Note: While this method will provide some rudimentary security (though pretty weak).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. for system 1 and system 2 respectively instead of just 6XXX and XX. If you have Dynamic IP addresses. as it requires an extension to be created for the peer Asterisk box. The above example assumes that both Asterisk boxes have Public Fix IP address. 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.g. com. Rather than being verbose. you must provide for security.org/book/export/html/1854 4/20/2011 . Unlike the first method. as this is pretty wide open. This method treats both the Asterisk box as internal to each other as peer and user. System 1 System 2 IAX2 Trunk Outgoing Dial Rules: 6XXX XX. As different installation resorts to different types of security arrangement. this second method will pass the Caller ID to the receiving party. this is simpler to set up.2 METHOD 2 . I will illustrate this method using tables as follows. I am using IAX2 for this purpose. The receiving party will actually get the callers’ extension number/ID instead of the extension number of the peer Asterisk box.au (or IP) Qualify=yes http://www. I will leave that to the individual implementer to deal with the security issues.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. however I believe. Note: You must provide for security. Trunk Name InterOffice InterOffice Peer Details host=xyz. 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 many ways.In a Peer/User arrangement Another method that I use is described below. This method does not require registration either and does not require you to create extensions for the peers. 27.freepbx. (Note: A little tutorial on DUNDi can be found here). Like all installation. 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.freepbx.org/book/export/html/1854 4/20/2011 .Administration Guide Page 13 of 42 type=peer host=def.com. all others go via City Office) Trunk Sequence IAX2/InterOffice IAX2/InterOffice Thinking of more than 2 boxes? http://www.com.(Apart from Local extensions.au (or IP) Qualify=yes type=peer User Context InterOffice-In InterOffice-In User Details context=from-internal host=xyz.com.au (or IP) type=user context=from-internal host=def. In this case. se same principle can be applied to more boxes. B and C (System 1. my advise to you is to hire a VOIP consultant. – with the appropriate dial plan of course. To provide a complete solution is beyond the scope of this document. Show below is just such a configuration. http://www. for example. Creating Administrator Roles Creating Administrator Roles For most web applications it is useful to have graduated permission access. so that users have only access to the functions they need. Mie is allowed to see status.And C peers with A.Administration Guide Page 14 of 42 Just as a matter of interest. while useable for a basic configuration. all external and inter-office (inter-branch) traffic goes via Box A. you can connect several boxes using this method.I believe. The following link will provide further reference for connecting two Asterisk boxes together http://www.B peers with A . While I have connected 3 boxes successfully. 2 and 3). Both the above methods. both Apache and iptables can be used to restrict access on a location basis to the web application. In addition to the webapp username / password settings. In my implementation I have box A.org/wiki/view/Asterisk+dual+servers If you require a complete solution tailored to your exact requirement.voip-info. Except for local traffic. access to the Extensions directory to change usernames and reset voicemail passwords as employees come and go. This lets you give office managers. Box A is the master box. All the other boxes use box A as the main exchange.freepbx. A good policy is to only allow local (LAN) or tunneled via SSH access to the web application. will not provide you with a complete solution. A peers with B and C . without exposing trunks and other settings they do not need. though exceptions can be be made for the Recordings (ARI) interface.org/book/export/html/1854 4/20/2011 . edit extensions (this part is not shown) and apply changes. If you know the room # of the guest you are trying to reach. 7. Press 1 http://www. Now upgrade the voice prompts to a paid voice or designated employee (the office manager or receptionist. press 2 for customer service. press 5 for office directions.) 9. 4. draw out on paper what you intend to to achieve. Welcome to BUSINESSNAME. Hospitality 1.press # to access the company directory. 3. ring groups. If you know the extension of the person you are trying to reach. Press 1 for sales. Please listen carefully as our options have changed. Office / Light industrial 1. Planning 2.org/book/export/html/1854 4/20/2011 . Show it to the customer. you should resist this impulse. 6. assigning a * feature code to whatever thing you want to test. Customer agreement with the plan. Record the audio prompts using System Recordings and an extension. 5. Normally heard as "Thanks you for calling MYBUSINESS. etc. day/night modes or time conditions). 2. for Service press 2". Welcome to HOTELNAME. press 3 for administration. First. or press 0 for the operator. Planning While the urge is strong just to dive in by clicking on IVR. etc. Bask in glory! Standard IVR Examples: 1.freepbx. for Sales press 1. Write out word-for-word what all the recordings are going to be. Run it by the customer (or your officemates). Please listen carefully as our options have changed. press 4 for Press inquiries. The proper flow to build a good IVR is: 1. Test all of these. you may dial it at any time. you may dial it at any time. 8. One way to do this is use miscellaneous destinations. Then go create your IVR. and then make the inevitable changes.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. Create any destinations that don't currently exist (queues. press 5 for Press inquiries. Engineering/Product Company with Direct Sales and Support 1. press 3 for technical support.1. If you know the extension of the person you are trying to reach. Welcome to BUSINESSNAME. press # to access the hotel directory.Administration Guide Page 16 of 42 for reservations. press 4 for hotel administration. press 6 for office directions. you may dial it at any time. or press 0 for the operator. press 5 for Press inquiries. Press 1 for sales. If you know the extension of the person you are trying to reach.org/book/export/html/1854 4/20/2011 . I strongly suggest you use an extension connected to the PBX to make your recordings. press 3 for event sales. press # to access the company directory. you may dial it at any time. 3. Retail 1. They'll be quick http://www.3. and directions. press 2 for customer service. Press 1 for sales. Shown here is 3. press 3 for store hours.freepbx. press 2 for customer service. press 2 for the front desk. or press 0 for the operator. 4. Please listen carefully as our options have changed. press # to access the company directory. press 4 for administration. press 5 for hotel directions. Making recordings Fire up the System Recordings module. locations.5. Please listen carefully as our options have changed. or press 0 for the operator. Welcome to BUSINESSNAME. press 4 for administration. spaces are not allowed in the names. For lame and silly reasons. When everything is all finished. If the recording is good enough (and don't obsess here yet).org/book/export/html/1854 4/20/2011 . or you'll get a cryptic error. Now that we've created a system recording. Creating the IVR When you select IVR.Administration Guide Page 17 of 42 and in the right format and you can worry about getting everything else right. To use your extension to make a recording. you can come back and replace those temporary recordings with paid or improved versions. 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. Dial *99 to listen to it. enter your extension in Step 1 and press Go. the first page is now a brief set of instructions on how to drive the IVR. we can create our IVR. We're going to start with a simple 1-level IVR . name the recording and press Save. You can either edit an IVR. Editing your IVR http://www. 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. or create a new one by clicking on 'Add IVR'. Don't skip this and go to Step 2. if one is existing. Now dial *77 and make your recording after the beep.freepbx. so the single Welcome-to-ACME recording will be enough. Enable Direct Dial: If you enable that. enter the option for the user. users will be able to dial the FeatureCodes">feature code for Directory. This may be one. This can be set to 'nothing'.You'll see it appear on the right straight away. Announcement: A System Recording that is played to users when they enter the IVR. this creates the IVR (and calls it 'Unnamed') as soon as you click 'Add' . Configuring your IVR In the box on the left. 'i' and 't' have special meanings: http://www. 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.Administration Guide Page 18 of 42 Unlike the old Digital Receptionist system.org/book/export/html/1854 4/20/2011 . or 't'. from the IVR and access the Directory service. users will. usually #. or. 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. be able to directly dial an Extension number.freepbx. in addition to being able to dial the IVR options. Advanced users can then use different IVRs to create a multi-tenant installation. or a series of numbers. Directory Context: This is the asterisk context of the directory. These are your options: Change Name: This is simply the descriptive name that appears on the right. 'i'. queues will not appear as a possible IVR destination if no queues exist. which is to play the menu three times and hangup. If you only have 1 2 and 3 defined. it will jump to this destination.freepbx. Creating and Assigning Extensions Creating and Assigning Extensions Numbering Schemes There are several schemes for assigning extensions. When you're finished. and caller pushes 4. click 'Save' and you have your new IVR. For example. you'll find the following http://www. A standard configuration is to go the operator. which is to play a 'invalid option' message and immediately replay the current menu. To test it. This won't let you decrease it to less than the number of options that are currently set. t: This overrides the default timeout behavior.G. 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.org/book/export/html/1854 4/20/2011 . though. Invariably. E. To delete an option. give it an incoming route or set up a miscellaneous application (* code) to reach it. simply leave the selection blank.Administration Guide Page 19 of 42 i: This overrides the default invalid choice behavior. to handle customers that don't have DTMFcapable phones. conf alarmreceiver. then extensions can be 200. this file must be done via the web gui. . 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/configuration_files . custom modifications.conf backup. 202.conf.org/book/export/html/1854 4/20/2011 . at minimum. 611 and 311 shouldn't be assigned. 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. etc. The basic rule is that all files are owned and modified by FreePBX unless they end _custom. to get the whole block of interest if possible.4.--------------------------------------------------------------------------------.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. as should other Miscellaneous Destinations Remember. but the rest of the 600s and 300s can be. All modifications to . So here is the list of files as of version 2.conf This file contains the crontab line(s) that will get executed for backup job scheduling. and it really hurts to run out.conf applications.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. choose the last 3 digits of the main number If the main number is 651-3200. Those owned by FreePBX will be in bold underline. It's usually low enough cost. For FreePBX. . this rules out extensions in the 100s and 900s. details at: http://freepbx. agents. There are alternative files to make .--------------------------------------------------------------------------------. There are a few exceptions to this rule but not many. . cdr_mysql. or emergency numbers In the US. If they become owned in a later version that version will be stated to the right of the file name. Don't collide with system shortcuts. 201. Do NOT edit this file as it is auto-generated by FreePBX. common dialing sequences. should be avoided. when reserving DIDs. . 7777 is commonly 'simulate an incoming call'. Be very careful as replacing an existing piece of code this way is the fastest possible way to break your system. then cdr_mysql will attempt to connect to the .conf (or extensions_additional. .conf please place it in extensions_override_freepbx.Administration Guide Page 21 of 42 . file. If hostname is not specified .conf) if so create that context in extensions_custom.if the database server is hosted on the same machine as the . 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. you place that code here as asterisk will only execute the first occurrences of that code and ignores other occurrences. extensions_custom.conf enum. extensions_additional. This file will not be overwritten.port=3306 . you can achieve a local Unix socket connection by .conf if you need to modify existing code code/context in extensions.org/book/export/html/1854 4/20/2011 .freepbx. This file will not be overwritten.conf) has a context or macro that you NEED to modify. .conf but read the notes about this file first. If hostname is specified . asterisk server.conf extensions.sock codecs.conf DO NOT EDIT THIS FILE. port specified or use the default port. setting hostname=localhost . it get's regenerated each and every time you apply changes. and additional code enhancements to the FreePBX dial plan.conf dnsmgr.conf this is the file that you place all your custom contexts. and is not "localhost". to the socket file specified by sock or otherwise use the default socket . If you are doing this you should probably think about filing for a feature request or bug fix to get it addressed properly. If you need to replace the functionality in extensions_additional. extensions_override_freepbx. port and sock are both optional parameters. then cdr_mysql will attempt to connect .conf please place your modifications in extensions_override_freepbx. [global] hostname=localhost dbname=asteriskcdrdb password=amp109 user=asteriskuser . or if hostname is "localhost".conf as asterisk uses the code for the first context referance and ignores additional occurances.conf extconfig. http://www.conf If extensions. Note . .conf dundi.sock=/tmp/mysql.conf and it will get called. freepbx.conf musiconhold_additional. http://www.conf manager_custom.conf iax_custom_post. Anything you can think of putting in this file can be placed into one of the _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.comf files where it will not get removed or replaced.conf features_general_additional. queues_additional.conf iax_custom.conf meetme_additional.conf manager.conf musiconhold.org/book/export/html/1854 4/20/2011 .conf parking_additional.conf iax_registrations.conf Do not edit this file in any way.conf file for your queues setup.conf iaxprov. queues_custom.conf files where it will not get removed or replaced.conf iax_additional.conf privacy.conf queues.Administration Guide Page 22 of 42 features.conf iax.conf meetme.conf musiconhold_custom.conf logger.conf oss.conf manager_additional.conf mgcp.conf phpagi.conf features_applicationmap_additional.conf features_general_custom.conf indications.conf Do not edit this file in any way.conf globals_custom.conf localprefixes.conf iax_general_custom.conf features_featuremap_custom.conf features_featuremap_additional.conf modem.conf features_applicationmap_custom.conf iax_registrations_custom.inc (should no longer be used as parking was moved to features) phone.conf iax_general_additional. Anything you can think of putting in this file can be placed into one of the _custom.conf modules. 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. If you need to adjust sip jitter or something else it will be sip_general_custom. Anything you can think of putting in this file can be placed into one of the _custom.conf.org/book/export/html/1854 4/20/2011 . create a context line: [79](+) then on the next line add the item(s) you need to add. sip_general_custom.Administration Guide Page 23 of 42 queues_custom_general.conf Do not edit this file in any way. This is the file that allows you to add or remove values to those entries found in the autogenerated queue_additional. Some of the required lines for nat'ing are externip=. it will get overwritten at some point and next time you restart your system you will suddenly wonder why things stopped working. 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).freepbx. etc.conf This is where FreePBX places all of it's general context settings.conf. So for example you have a queue 79 that need a additional parameter added. see sip_general_custom. queues_general_additional. To remove use (-) instead followed by the line(s) you want removed.conf.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 sip.conf.comf files where it will not get removed or replaced. 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 want to add additional setup parameters for your sip device see sip_custom_post.conf for more info. If you have a legacy system these lines might have been placed in sip_nat. nat=. If you are looking to do nat'ing.conf rtp.conf Do not edit this file in any way. See sip_nat. If you need to override one of these or add a new one please do so in sip_general_custom.conf file.comf files where it will not get removed or replaced. and optionally fromdomain=.conf or if it is a legacy system sip_nat. Anything you can think of putting in this file can be placed into one of the _custom. queues_post_custom.conf in the past.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. sip_general_additional. res_mysql. http://www.conf (if it is for the general context) or sip_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 do edit this file and place something new in it. localnet= (you can have more then one occurrence of this line). To remove use (-) instead followed by the line(s) you want removed.255. sip_notify.168.0 subnet Requires these two lines in the either sip_general_custom. sip trunks.255.conf or sip_nat. 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). If you move the lines from this file to sip_general_custom.2 on a 192. Create a context line: [1000](+) then on the next line add the item(s) you need to add. 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.conf This is the first file that is not under the general context.255. sip_additional.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.168.conf http://www.168. trunk.0 sip_nat. sip_custom.conf file localnet=192.conf General section registrations that are auto-generated by FreePBX. see sip_custom_post.255.255.conf.255.conf.1.255.0/255. Example: Server 192. The new preferred location is sip_general_custom.0/255.0/255. sip_registrations.0 network Phones inside the office are on the 192.conf This is where FreePBX puts all sip extensions.conf a custom file just in case there is ever a need to override a general registration that was autogenerated by FreePBX.conf file. IT allows you to define contexts that you need before the contexts that are auto-generated by FreePBX in sip_additional. etc. If you need to add a additional parameter to a extension.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.0 localnet=192..168.conf This is the file that allows you to add/remove values to those entries found in the auto-generated sip_additional.1. sip_registrations_custom. So for example you have an extension 1000 that needs an additional parameter added.conf This is the old common location for placing the lines needed to enable the nat'ing of SIP.conf. sip_custom_post.0/255.168.org/book/export/html/1854 4/20/2011 .2.1.255. etc.freepbx.2. inc #include vm_email. These lines will be generated by FreePBX every time you add/edit/delete a extension.inc file. 99% of the world needs to edit two lines in the vm_general.inc http://www. vm_general. so please be careful.inc this file contains the e-mail subject line and message body for any voice mails that are e-mailed. 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. [general] #include vm_general.conf This file is both editable by you and by FreePBX. If you need to edit the mail sending parameters edit the vm_general.inc file at the initial build time. The most common change to this file is to create a context called [zonemessages]. The structure of this file is as follows: [general] #include vm_general.inc #include vm_email.inc file.conf voicemail.Administration Guide Page 25 of 42 skinny.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.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. If you create this context it should be placed after the second #include line and before the [default] line.inc [default] Once you have configured a system with voicemail there will be values after the context [default].org/book/export/html/1854 4/20/2011 . 26MB Download.org/book/export/html/1854 4/20/2011 . 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. High Quality Sounds For those using the sounds that come with asterisk. zapata. Berkeley nor the names of its contributors may be used to endorse or promote http://www. to have all of the asterisk sounds re-recorded.) the system acheives. maxmsg= limits the total number of messages allowed in a mailbox. Redistribution and use in source and binary forms. 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 . Kristian Kielhofner of astLinux has come to the rescue by paying. Neither the name of the University of California. etc. this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. are permitted provided that the following conditions are met: Redistributions of source code must retain the above copyright notice. Make sure and note what call levels (and conferences.conf zapata_additional. Redistributions in binary form must reproduce the above copyright notice. out of his own pocket. 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. with or without modification. due to them only being in GSM format. this list of conditions and the following disclaimer. other common lines to edit are: maxmessage= this is the max message limit. and has released them under the BSD-License for all to use. you'll know that volume and timing can be a bit wonky sometimes.conf zapata_custom_chan_default.conf Hardware examples Add child pages to enter hardware examples here. and the quality isn't all that great.freepbx.Administration Guide Page 26 of 42 this file contains the e-mail / voice mail configuration parameters. Interfacing to a PSTN 9. enter the following from the command line. PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES. you will need to edit the following files. 4.freepbx. STRICT LIABILITY. 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. 1. You need to edit this. IN NO EVENT SHALL THE REGENTS AND CONTRIBUTORS BE LIABLE FOR ANY DIRECT. OR CONSEQUENTIAL DAMAGES (INCLUDING. If you have this card installed. zapata. 5. INCLUDING. 3. THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES.conf.org/book/export/html/1854 4/20/2011 . OR PROFITS.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). LOSS OF USE.x http://www.conf and modules. BUT NOT LIMITED TO. INDIRECT. BUT NOT LIMITED TO. it may be necessary to configure it by using the zaptel card auto-config utility so the correct zaptel driver will be set up. zaptel. EXEMPLARY. SPECIAL. DATA. 2.conf for AAH 1. OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE. 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. EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.Administration Guide Page 27 of 42 products derived from this software without specific prior written permission. If this card is added after Asterisk has been configured. To make outbound calls you will need to set an outbound route as well. WHETHER IN CONTRACT. OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY. THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. INCIDENTAL. 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. To do that. add this line http://www.conf.2 zaptel.0 (you may have to experiment a little with this setting) txgain=8.org/book/export/html/1854 4/20/2011 .conf Change the loadzone and defaultzone to 'au' # Global data loadzone = au defaultzone = au 9. alias char-major-196 torisa options wcfxo opermode=AUSTRALIA .Include AMP Configs channel => 1 #include zapata_additional.1. 9. add the line highlighted in Bold below: .freepbx.1.conf Leave the rest of the file as it is. It is located at the end of the file. 9.1 zapata.Administration Guide Page 28 of 42 or modprobe. The last 2 files live in the /etc directory – use a text editor to edit them.conf for AAH 2.1.conf (modprobe. I have also changed the following setting to obtain a good compromise on volume/echoing: rxgain=10. .x.0 (you may have to experiment a little with this setting) Ensure the following exist in zapata.3 modules.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.x.x) For AAH 1.conf Under [channels] edit the following lines: [channels] busydetect=yes busycount=6 For my installation to function correctly.x. 1 zapata-auto. Set them up as per setting up routes in the earlier chapters of this document.this is a trunk. If this card is installed after Asterisk has been loaded. reboot your PC and when Asterisk starts. use AMP to add a route for incoming calls or asterisk will not answer your trunk.Administration Guide Page 29 of 42 install tor2 /sbin/modprobe --ignore-install tor2 && /sbin/ztcfg . Once this is done. Span 1: WCTDM/0 'Wildcard TDM400P REV E/F Board 1' signaling=fxo_ks .Note . Similarly. Create a ZAP trunk in AMP for Channel 2 context=from-pstnchannel => 2 < . it will look something like the illustration below (see the red highlight) zapata-auto.conf file. this card has 4 module ports that can be loaded with FXS or FXO modules. Note .conf . 9. When you open the zapata_auto.conf file and you will see a list of all your channels in your TDM400P.2 DIGIUM TDM400P FXO/FXS CARD Like the Digium Wildcard X100P. http://www. Set up the trunks as trunks and the extensions as extensions in AMP. this card allows you to connect a POTS (plain Old Telephone System) line to your Asterisk@Home box.2.conf Next.freepbx. Create a ZAP extension in AMP for Channel 1 channel => 1 < . to make outbound calls you will need an outbound route.-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.this is an extension. you will need to configure it just like the X100P by using the following command on the command line: genzaptelconf 9.org/book/export/html/1854 4/20/2011 . Channel 1 is the top RJ-45 on the back of the TDM400P card. Unlike the X100P. look in the zapata-auto. using config edit.-this would have been defined already by the config signaling=fxs_ks . Or.org/book/export/html/1854 4/20/2011 .conf.x) You will need to edit the modules.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 modules.conf for AAH 2. zapata.--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.1.aumixrc -L >/dev/null 2>&1 || : pre-remove sound-slot-0 /bin/aumix-minimal -f /etc/. options wctdm opermode=AUSTRALIA fxshonormode=1 bootstringer=1 Your modules.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.2. in AAH Ver.2.--ignore-install http://www.conf as per the X100P card in the previous section.freepbx.conf and zaptel.conf (modprobe. 9.conf to add the necessary option for usage in Australia. or modprobe. you may also do the following: Locate the line 'install wctdm /sbin/ztcfg-. Do not change anything else. The example below is for AAH 1. where you need to add the following line.x. you will need to edit the following files.conf (AAH 1. 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.6.4.freepbx.2.h spinlock. fxshonormode=1' Also see Appendix E. by selecting opermode=AUSTRALIA the zaptel drivers automatically add the 'boostringer=1 .3 (Users Suggestions) 9.h.3 REBUILDING ZAPTEL DRIVER Every time there is a kernel update with yum (which is the case with Asterisk and CentOS).com/index.org/book/export/html/1854 4/20/2011 . Unfortunately. ztdummy if you don’t have any ZAP devices. The following is the fix .h Once the file has been retrieved.php?p=123 Log into your new server as root and issue the following commands: cd /usr/src/kernels/2. ZAP device support needs to be rebuilt using the new kernel.source Nerd Vittles http://nerdvittles.com/aah27/spinlock. this will cause a slight problem as RedHat bug caused the rebuilding process to fail. reboot using the following command: shutdown -r now When the reboot completes.EL-i686/ include/linux mv spinlock.Administration Guide Page 31 of 42 wctdm && /sbin/ztcfg Note: as of Zaptel Drivers 1.old wget http://nerdvittles. you can start rebuilding the support for your ZAP devices or for that matter.9-34. Take another snapshot for good measure.Administration Guide Page 32 of 42 ..36 Regional tab http://www. then upgrade it to the latest version (at the time of writing.2 Change the settings System tab DHCP: No Static IP: something on your local subnet e. Now reset SPA-3000 back to factory defaults.g.g. 203.12.200 NetMask: 255.255.5a). 192.g.1. 9.12.. 9.1 Log in to SPA3000 Login to your SPA-3000 as admin/advanced.35 Secondary DNS: your ISP's secondary DNS address e.4 SIPURA SPA3000 AS A PSTN INTERFACE To those new to the SPA3000.freepbx. as no one single source of information that I've found so far has a config that would actually work for me. Nothing should have changed in your settings. (See also user Users’ Suggestions) 9.4..4.org/book/export/html/1854 4/20/2011 . in case you ever want to know what the defaults were.. there is a simplified installation and configuration instruction by JMG Technology.160.html page) of your current SPA-3000 configuration.255. it gives a good insight of the Sipura SPA3000’s capabilities.254 Primary DNS: your ISP's primary DNS address e. 203.168. If you're not already running the latest SPA-3000 firmware.1.0 Gateway: your router's IP address e. Before you change anything.e. except that you have a few extra options that you didn't have before. Take another snapshot now too. just in case you ever need to refer back to your own customisations.g. To help them in their endeavours. I've put the following together. because I'm only going to list the minimum changes required to keep things simple.and you're done. it's 3. I'd suggest taking a snapshot (i. just save the .168. While it is directed mainly at standalone ATA users. 192.160.. 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.1. etc.4/2.2...4/.2..g.4/2.2/.2.4/.. 192.2.4/.4/2..10(..4/2.2/1+2+3.5/3.234 Register Expires: 60 Display Name: Whatever User ID: Asterisk extension number e.10(.freepbx...4/.4/2....2/1) Ring Back Tone: [email protected]/2.2/..10(*/0/1+2) Busy Tone: [email protected]@[email protected]@-19.5/3.4/.07 Hook Flash Timer Max: .Administration Guide Page 33 of 42 Dial Tone: [email protected]/2/0) Ring 1 Cadence: 60(1.. 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.*(.org/book/export/html/1854 4/20/2011 ..4/.4) Ring 3 Cadence: 60 (1..4/.4/.2. |[4689]xxxxxxx|7777|899060xxxxx.13 Delete all the Vertical Service Activation Codes.2...2/1+2+3.2.4/2) CWT8 Cadence: 30(..4/2.4/.|x.168.2.2.4/.4. FXS Port Impedance: 220+820||120nF Line 1 tab Proxy: IP address of your Asterisk box e.4/2.4/1) Reorder Tone: [email protected]) Hook Flash Timer Min: .4/2.|0[23478]xxxxxxxx|09xxxxxx|1100 |122[135]|1222xxxxxxx|12510[12]|12554|1[38]00xxxxxx|13[1-9]xxx |1747xxxxxxx|2xx|393xxxxxx|3xxxx. but I like to do a bit of sanity checking..) for example (*xx..4/.2.2.. PSTN Line tab (method 1) http://www.) will work.4/.4/.. g.Administration Guide Page 34 of 42 Proxy: IP address of your Asterisk box e.org/book/export/html/1854 4/20/2011 ..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. 192.168.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.freepbx.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.1 Disconnect Tone: [email protected]@-30. which I am currently using.org/book/export/html/1854 4/20/2011 .Administration Guide Page 35 of 42 PSTN Hook Flash Len: ..1(. PSTN Line tab (method 2) Proxy: IP address of your Asterisk box e.1. 192.g.375/. 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.1(.org/book/export/html/1854 4/20/2011 .g.375/.3 Add SIP Trunk Then in [email protected]. Outbound Caller ID: <0298765432> (for example) Maximum Channels: 1 Dial Rules: 0+NXXXXXXXX (for example) 0011+ZXXXXXXXXXX. (S0<:s@192. User 1 tab Default Ring: 3 Default CWT: 8 9.4.375/1+2) FXO Port Impedance: 220+820||120nF On-Hook Speed: 26ms (Australia) Using this alternative method.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. You may also get CLID if your Telco has activated incoming Caller ID on your phone.freepbx. Trunk Name: telstra (for example) Peer Details: canreinvite=no http://www.Administration Guide Page 36 of 42 Dial Plan 1: (S0<:s@YourAsteriskIP>) e.168.1 Disconnect Tone: 425@-30. 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.Administration Guide Page 37 of 42 context=from-pstn host=the IP address of your SPA-3000 (for example.freepbx. 192.200) 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).1.1. (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. 192.168.168.4 in Sipura SPA-3000 http://www.org/book/export/html/1854 4/20/2011 . which goes to your chosen Destination. you will be somewhat disappointed. amongst others.freepbx. also play very important roles.00 from Dick Smith . Your only other initial cost will be the $20. Ensure that the PC has an Ethernet NIC for connecting to your home network. if you want the ability to make PSTN calls. What is it going to cost? Assuming that you already have a broadband service. a router. If you want to restrict all your calls to VOIP only. but if you will be happy with a quality that is not quite but close to your existing PSTN calls. you might be in luck. Astratel. which may include a monitor. and a Windows PC to run the softphone. If you do not have a spare PC with the above specification. Linux CLI Commands Entering the Asterisk Console http://www. then you may be able to buy one from your local swap meets for under $200. Spantalk etc will register you for SIP communication for free provided that you do not need to make PSTN calls. All these “Major Expenses" will be recovered when you receive your monthly Telstra or Optus phone bills.Australia.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. it may not cost you anything at all. If you already have a spare computer to dedicate to this task.00. VOIP via the Public Internet is very much dependant on a number of factors – available bandwidth not withstanding.org/book/export/html/1854 4/20/2011 . then the cost is almost nothing unless you need to buy an audio headset ($15. Some economic and quality considerations should be examined. your usage habit of the internet and LAN traffic and equipment quality. the cost will be minimal. What will the Quality of the phone calls be? If you are expecting the quality to be as good as your existing PSTN calls.00 or so activation fee to Oztel (or other VSP of your choice). Some VSPs like Pennytel.) for the softphone. freepbx.org/book/export/html/1854 4/20/2011 .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.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. Choose a password that is at least 4 digits. 2 for my assistant or just leave a message after the tone. Common options are 'press 1 for my cell. System Tools Area for additional system tools for Asterisk and FreePBX http://www. Don't send from dynamic IP addresses. Voicemail pager feature This gives a short description of the message envelope. Use dyndns.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. as matter of policy. Set your hostname to be a fully-qualified domain name. 2. Voicemail Locator (VMX) Feature This optional feature lets users set up a short menu before voicemail takes the actual message. the following are recommended: 1. suitable for emailing to a wireless carrier's email gateway. and directions to change it. Instructing New Users New users should. their voicemail initial password. Voicemail in email feature In order for these emails to pass through spam filters.' Resetting It's commonplace to have to reset the passwords as people leave the company. Assigning Voicemail PasswordsYou must enter a voicemail password when creating an extension enabled with voicemail.freepbx.org/book/export/html/1854 4/20/2011 . Resist the impulse to standardize the default. and the system will be insecure.org if you don't want to pay for a real one. be sent an email with both general instructions. sourceforge.net/sourceforge/webadmin/webmin-1. etc.101:10000 To update WebMin Anytime you want to update Webmin.2601. It is written and maintained primarily by Simon Tatham and can be downloaded from the following link.2601.168.noarch.html WEBMIN WEBMIN Webmin in an invaluable web based gui for managing a Linux box.dl.com/support/solutions/ydl_general/webmin. However there are some users who found that following an alternative method is simpler.rpm I have found the above method is straightforward and simple.Administration Guide Page 41 of 42 Putty PuTTY PuTTY is a free implementation of Telnet and SSH for Win32 and Unix platforms. If that is the case.noarch.putty.org/book/export/html/1854 4/20/2011 .noarch.g.shtml You may connect to Webmin remotely through your browser using the following address http://<YourAsterisk_IPAddress>:10000.nl/download.sourceforge. simply do the following. Those who want to use Web Admin to maintain the Asterisk System may download Webmin from here or from CLI.dl. do the following: wget http://superb-east. rpm –Uvh http://superb-east.rpm Or be totally lazy like me and do the whole lot in a one liner. editing files.net/sourceforge/webadmin/webmin-1.terrasoftsolutions. E. Webmin make it easy to configure application like SMTP mail. 192. system settings.rpm Install it with the following command through CLI: rpm -Uvh webmin-1.freepbx.0. along with an xterm terminal emulator. the alternative installation method can be found here: http://www. http://www. http://www.260-1. Administration Guide Page 42 of 42 Log on to your Asterisk box (SSH or at the console). Legacy SCP protocol is also supported. Its main function is safe copying of files between a local and a remote computer.php http://www. http://winscp. At the command prompt.org/book/export/html/1854 4/20/2011 .freepbx. 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.
Copyright © 2024 DOKUMEN.SITE Inc.