Getting Started

From PBXes

(Difference between revisions)

Current revision

1 Definitions
- 1.1 Extensions
- 1.2 Trunks
2 Adding extensions
- 2.1 Classical extensions
- 2.2 SIP phones
3 Adding trunks
4 Choosing routes
5 Troubleshooting

[edit] Definitions

[edit] Extensions

An extension is just the definition of a classic (POTS) or SIP (VOIP) telephone line/number. Inbound calls can be routed to either of these two extensions types using inbound routes. Classic extensions will be dialed using the defined outbound route definitions. SIP extensions are normally used to register SIP devices to, like SIP phones, softphones or ADSL modems having SIP capabilities. Unlike classic extensions, SIP extensions can be used to place calls too.

[edit] Trunks

Trunks are definitions of the SIP providers you want to use for receiving inbound calls and/or placing outbound calls. Inbound calls are routed to extensions using inbound routes, and outbound calls are routed to a trunk using the outbound routes. For receiving inbound calls it is necessary to register to the SIP provider.

[edit] 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.

[edit] 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 and 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.

[edit] 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
Registration Interval	Fixed IPs: 1 hour Dynamic IPs or with issues: 2 minutes
Port for UDP and TCP	5060
Alternative Ports for UDP and TCP	53, 69, 80, 135, 161, 443, 500, 1433, 1701, 1812, 3389, 4500, 5061, 5900, 16999, 26999 and 36999 (recommended) Note: Because of DNS entries for pbxes.org your device may be selecting port 5060 automatically. If you want to use an alternative port enter 144.76.38.78 as SIP server.
Port for TLS	5070

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 (*97) 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)

[edit] 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 or PRO 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 landline can also be forwarded to your SIP DID.

[edit] Choosing routes

To get inbound and outbound calls working properly, you have to add at least an Inbound Route and an Outbound Route.

When entering your first inbound route please leave the fields Trunk and Caller ID Number blank to create a general rule.

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 and PRO Accounts also allow to choose a sequence of trunks for having a backup if a trunk is busy or goes offline.

[edit] Troubleshooting

SIP calls over 3G networks require better signal strength than regular voice calls. Therefore we recommend to call over SIP only if you have 3 or 4 bars while you are not moving, except in downtown areas where coverage is always 3 or 4 bars.

Dropouts on softphones can be caused by other apps initiating data transfers at the same time, or consuming CPU power. If there is no late or lost packets shown in statistics, try to switch off screen to improve.

As the next steps further reading of pages FAQ and Fixed Mobile Convergence (FMC) is recommended.

Languages:

English • Deutsch

Retrieved from "http://345946.n0h2l.group/wiki/index.php/Getting_Started"

 Dynamic IPs or with issues: 2 minutes
 |-
-| Remote Port	|| 5060
+| Port for UDP and TCP || 5060
 |-
-| Alternative Ports || 53, 69, 80, 135, 161, 443, 500, 1433, 1701, 1812, 3389, 4500, 5061, 5900, 16999, 26999 and
+| Alternative Ports for UDP and TCP || 53, 69, 80, 135, 161, 443, 500, 1433, 1701, 1812, 3389, 4500, 5061, 5900, 16999, 26999 and
 (recommended)
 Note:
-Because of the DNS entries for pbxes.org your device may be selecting port 5060 automatically. If you want to use an alternative port enter 188.40.65.148 as SIP server.
+Because of DNS entries for pbxes.org your device may be selecting port 5060 automatically. If you want to use an alternative port enter 144.76.38.78 as SIP server.
+|-
+| Port for TLS || 5070
 |}
 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:
+The users may now sign in to listen to their voicemail (*97) or display the call details which pertain to their extension only. Provide them with the following info:
 {| border="1" cellspacing="0" cellpadding="3"
 == Troubleshooting ==
-SIP calls over 3G networks require a better signal strength than regular voice calls. Therefore we recommend to call over SIP only if you have 3 or 4 bars and you are not moving, except in downtown areas where coverage is always 3 or 4 bars.
+SIP calls over 3G networks require better signal strength than regular voice calls. Therefore we recommend to call over SIP only if you have 3 or 4 bars while you are not moving, except in downtown areas where coverage is always 3 or 4 bars.
 Dropouts on softphones can be caused by other apps initiating data transfers at the same time, or consuming CPU power. If there is no late or lost packets shown in statistics, try to switch off screen to improve.
+As the next steps further reading of pages [[FAQ]] and [[Fixed Mobile Convergence|Fixed Mobile Convergence (FMC)]] is recommended.
 {{Languages|Getting Started}}