24.0.1
The background of SIP TrunkBackground on SIP Trunks
The SIP Trunks serve are used for connecting PrivateServer to a local PBX or to a SIP Voice VoIP provider. The above statement implies that a A Trunk can be used both for receiving and placing calls in a bidirectional way.
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| By Secure Call design the SIP Trunks are available only for the trunks are mainly useful to interconnect EVSS ecosystem with other SIP infrastructure, using an end-to-site security model, featured by PrivateGSM Enterprise Edition. SIP trunks have a very limited usefulness for PrivateGSM Professional Edition can't place calls on a Trunk. |
The PrivateServer is designed to perform always as a client when configuring a SIP Trunk. This means that PrivateServer itself can not act as a Trunk Provider.
Even if a SIP Trunk can receive and place calls PrivateServer has to be configured to act in a bidirectional way. The Trunk setup should be done for the incoming calls first and after this function is enabled you can setup it for outgoing calls as well.
The incoming calls Trunks are configured in the Inbound entry that you access under Main Menu -> PBX INTEGRATION. Outgoing calls are configured into the Outbound entry that is under the Inbound one.
2.2 Inbound Configuration
Click on the "Inbound" link in menu on the left side to go to the "Sip Trunk List" page.
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| SubtitleText | The "Sip Trunk List" page |
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| AnchorName | empty_trunk_list |
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| AnchorName | empty_trunk_list |
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illustrates this page, with an empty table and a "New Sip Trunk" icon above it. This is the default empty configuration.2.2.1 Create an Inbound Trunk
Click on the New Sip Trunk to go to the Edit Sip Trunk page:Each SIP Trunk can be used to both receive and place calls.
4.0.2 SIP Trunk's categories
From the PrivateServer version 2.5 you have two main SIP Trunk's categories:
- Secure Trunks
- Insecure Trunks
The former ones entail both TLS and SRTP, while the latter none of them: they use just RTP over UDP (thus without TLS). Which one is to be used depends on many factors, major one is the kind of PBX on the trunk's end. For example Cisco Unified Communications Manager are usually connected to PrivateServer via Secure Trunk, while SIP providers most often use the Unsecure Trunk.
4.0.3 Authentication Models in SIP Trunks
Apart for trunk's categories, you should get acquainted with the Authentication Model, that can be:
- SIP Account for Registered Trunks
- IP Authenticated forUnregistered Trunks
SIP Trunks using the Registered Authentication Model need an account on the other end. Then they use the account's credential to authenticate the Trunk when the PrivateServer is started and every communication that passes over that Trunk is considered authenticated, thus valid by default, and has the right to be processed and routed.
SIP Trunks with no registration, on the other hand, do not need for an account to be setup: instead they use IP address to authorize every SIP communication directed to the PrivateServer. This model implies that any new SIP communication is authenticated against the sender's IP address (meaning the one belonging to the PBX connected to the PrivateServer).
The latter model is mostly used enterprise systems, eg with CUCM. The former is mostly used with SIP Providers.
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There's no way to perform an unauthenticated SIP INVITE on PrivateServer! You can have different authentication models but you cannot choose to enable an unauthenticated SIP Trunk. |
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4.0.4 General starting point for SIP Trunk configurationNo matter what kind of Trunk you're going to configure on PrivateServer, you would anyway come throughout the SIP Trunk configuration page in order to complete your setup.
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| SubtitleText | "Edit Sip New SIP Trunk " form |
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| AnchorName | edit new_sip_trunk form |
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There are two types of trunk that you can create, depending on the authentication type: - username and password: you must insert username and password
- IP based: leave the username and password fields blank and fill in the HOST parameter.
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The fields have to be configured as follows:
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in
it's shown the tipical generic new sip trunk form. The fields have the following meanings:
- NAME: a meaningful name for this trunk
- HOST: IP address or Hostname of the SIP server provided by ITSP
- OUTBOUND PROXY: IP address of the outbound proxy server provided by ITSP
- VIRTUAL PHONE NUMBER: associated virtual number
- USERNAME: login username
- PASSWORD: password for the username entered above
PORT: registration port of the service- REGISTER: select if you want to have only outgoing calls or incoming (see below for deeper explanation)
- SEND KEEP ALIVE: select it if you want PrivateServer to send a SIP OPTIONS packet on a regular basis
- PORT: registration port of the service
- TRANSPORT: protocol to be used as a transport mean. Choices are:
- ENCRYPTION: whether the trunk has to be encrypted or not
- MAX CONCURRENT CALLS: maximum number of calls on a single trunk
- NAT: enable/disable the Network Address Translation configuration
- DIRECTMEDIA: experimental sound management in P2P mode without routing the audio stream through the server
- SENDRPID: the Remote Party ID, used to interconnect with VoIP Provider for the management of privacy
- AUDIO TONES: this is used for Audio Messaging. The possible values are:
- NO AUDIO TONES: no audio messages would be played on this trunk
- ON EARLY MEDIA: the audio messages would be played using SIP early offer
- ON ANSWERED CALLS: the audio messages would be played using SIP delayed offer
- DTMFMODE: tone signaling technology used for this SIP interconnection
- ALLOW: voice codecs used in this trunk
- DISALLOW: voice codecs denied in this trunk
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The called number in inbound calls passing through the trunk should match the number of an internal account or a virtual phone number associated with an account. This way the correct SIP phone will ring. |
The default values for Port, Allow and Disallow are usually work well, however feel free to change them to suite your needs better.
Mandatory fields are:
Register option
If the trunk is going to be also used for OUTBOUND calls (external calls) you need to be sure that the "Register" checkbox is checked. In this case the "Name" and "Secret" fields are mandatory as well. Otherwise you can uncheck the "Register" checkbox so the above fields will become optional.
Fill out the fields accordingly to the guidelines above and when you are done click on the "Update" icon at the bottom of the page.
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| SubtitleText | The new SipTrunk List TableSIP Trunk section in the Main Menu |
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| AnchorName | new trunk list |
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You can get back to the "Sip Trunk List" page where the Sip Trunk table is now filled based on your settings ( |
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You can reach
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| AnchorName | new_sip_trunk list |
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).A notification will inform you about the progress of the operation. The number shown in the notification is an internal identifier in the database and thus can be ignored.
To check the entered data of the SIP Trunk click on the "Host" value into the table (for example in this case you should click on "10.0.0.101"). This will open again the Edit Sip Trunk page with the previously entered data.
2.2.2 Change Inbound Trunk settings
On the Edit Sip Trunk page you can change the values of the Inbound Trunk fields. When you are done commit the changes by clicking on the Update icon.
Click again on the "Host" value and verify the values of the fields are actually changed.
2.2.3 Delete an Inbound Trunk
To delete a trunk you need to select it in the trunk list. Click on the selected trunk and you'll be taken to the Edit Sip Trunk page. Click on the Delete icon at the bottom of the page to delete the Sip Trunk.
using the PBX INTEGRATION menu inside the main menu. As you can read in | Xref |
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| AnchorName | pbx_integration_menu |
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there are just 2 links inside it:| Subtitle |
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| SubtitleText | The Sip Trunk List after the Trunk deletion SIP Trunk list |
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| AnchorName | sip trunk deleted |
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A warning pop-up window will ask you for confirmation. After you click on the Ok button or press the Enter key on your keyboard you'll be taken back again to the "Sip Trunk List" page which will show an empty list and a notice as in | Xref |
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| AnchorName | sip trunk deleted |
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. 2.3 Outbound Configuration
Click on the "Outbound" link in menu on the left side so you get to the "Dialling Rule List" page. Here you can choose the default outbound trunk or set up the dialling rule to route the calls on a specific one.
2.3.1 Set the default Outbound Trunk
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| SubtitleText | Default Outbound view |
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| AnchorName | basic outbound view |
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As soon as the page is presented to you, you can see it as in
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| AnchorName | basic outbound view |
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. The upper side is about the Default Trunk used for outbound calls. By default any outbound call is Blocked and this behaviour stills until a SIP Trunk is chosen.| Subtitle |
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| SubtitleText | choose the default outbound trunk |
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| AnchorName | choose default outbound |
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To choose the default Trunk you just need to click on the "BLOCKED" tag and then you can use the drop down menu as in
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| AnchorName | choose default outbound |
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If no Inbound Trunk has been configured, then the drop-down menu is empty. |
Once you're done you can press the "Update" button to confirm your choice. You return to the "Dialling Rule List" page and this time the "Default Outbound" option is set to the trunk you choose.
2.3.2 Create Outbound routing by dialling rules
The Outbound Trunks can also be selected by dialling a prefix. If you need to create conditional routing rules for Outbound calls, then you need to use the lower section of the "Dialling Rule List" page (cfr.
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| AnchorName | basic outbound view |
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| SubtitleText | The "New Dialling Rule" button |
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| AnchorName | new dialling rule link |
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After you press the "New Dialling Rule" button (which is shown in
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| AnchorName | new dialling rule link |
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) you are redirected to the "Create Dialling Rule" form (shown below).| Subtitle |
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| SubtitleText | "Create Dialling Rule" form |
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| AnchorName | create dialling rule |
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2.3.2 Read Outbound routing by dialling rules
If your need is to set up a distinct rule
2.3.2 Update Outbound routing by dialling rules
If your need is to set up a distinct rule
2.3.2 Delete Outbound routing by dialling rules
If your need is to set up a distinct rule
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For creating a new SIP Trunk you go on the Inbound menu voice and then use the New Sip Trunk in the Sip Trunk List page (see
.For each category and authentication models you are going to find a specific manual page describing the correct configuration and the possible parameters to be used.