Getting Started
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| | Domain/Realm/Server || pbxes.org | | Domain/Realm/Server || pbxes.org | ||
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| - | | UDP Port || 5060, 5061 or 80 | + | | UDP Port || 5060, 5061, 443 or 80 |
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Revision as of 20:43, 2 November 2007
Contents |
Adding extensions
Select "Extensions" from the menu to add two kinds of extensions. Existing phone numbers attached to PSTN lines, as well as IP phones, ATAs and soft-clients to your PBXes account. Extension numbers may be 3 or 4 digits. Avoid using 2 digit extensions, since they may conflict with Ring group numbers.
Classical extensions
Classical extensions are used to reach PSTN numbers, which by definition cannot be reached via SIP URIs. These can be existing land-lines in the office at home or mobile phones. Just enter an extension number 3 or 4 digits, a name and the telephone number of the land-line.
Classical extensions can only be called by using a trunk, so set an Outbound route appropriately in order to reach them.
SIP phones
In addition to extension number and name enter a password. The following parameters are assigned to the SIP phone to register with the PBX:
| Login | <account name>-<extension no.>
e.g. pamsoft-200 |
| Password | <extension password> |
| Domain/Realm/Server | pbxes.org |
| UDP Port | 5060, 5061, 443 or 80 |
To verify the settings dial *43 on the IP phone to perform an echo test. In case the key tones are not recognized during a call, try setting a different "dtmfmode". The allowed values are rfc2833, inband, info and auto.
The users may now sign in to listen to their voicemail or display the call details which pertain to their extension only. Provide them with the following info:
| Login | <account name>-<extension no.> |
| Password | <voicemail pin> (4 digits) |
Adding trunks
Next select Trunks from the menu to enter the SIP credentials provided by your ITSP, so your PBXes can communicate with the PSTN world.
SIP providers supply both the incoming (DIDs) and the outgoing trunk lines to your PBXes. Select providers which use the SIP protocol and usually require a username and password to access their service. Before going on to the next steps test the SIP account for outgoing calls from your PC with a soft-phone.
Holders of a Premium Account can see our recommendation of tested SIP providers on the Trunks page.
SIP providers may supply a phone number (DID) through which your PBXes account can be reached on. Depending on the provider this may be a local number in a particular country. In certain countries it is not yet possible to port a PSTN number to another provider (LNP). If you want to keep your PSTN number and integrate it with your PBXes account, consider connecting and configuring a VoIP gateway for your existing PSTN line. The calls to your existing land=line can also be forwarded to your SIP DID.
Choosing routes
To get inbound and outbound calls working properly, you have to configure the items Incoming Calls, Inbound Routing and add at least an Outbound Route.
To add an Outbound route for all destinations enter "XX." as a dial pattern or pick "All" from the predefined dial patterns. This is not the recommended procedure to setup your Outbound routes, it is only intended as a quick verification your trunks for outgoing calls work properly.
If there are several trunks you may choose the destination numbers to select the trunks. Outbound routes are processed from top to bottom. Routes can be reordered by clicking on the small arrows next to the route names. When all the changes are finished, clicking on the red bar puts the changes into effect.
Premium Accounts also allow to choose a sequence of trunks for having a backup if a trunk is busy or gets offline.
