Grandstream CloudUCM
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| - | [[File: | + | [[File:VoIPms Grandstream CloudUCM Logo.jpg|frameless|right|312px]] |
| - | CloudUCM is a cloud PBX that provides a scalable business communication and collaboration platform with powerful features and integrations that enable teams to be more productive than ever before. | + | '''[https://www.grandstream.com/products/ip-pbxs/ucm-series-ip-pbxs/product/clouducm Grandstream CloudUCM]''' is a cloud PBX that provides a scalable business communication and collaboration platform with powerful features and integrations that enable teams to be more productive than ever before. |
| - | + | == Configuration == | |
| + | === Logging in === | ||
| + | [[File:VoIPms Grandstream CloudUCM Login.jpg|frameless|right|400px]] | ||
| + | You can log directly to your instance URL or access it from your [[Grandstream Device Management Cloud System (GDMS)]] account. | ||
| - | == | + | === Creating a trunk === |
| + | [[File:New SIP trunk.png|border|right|576px]] | ||
| + | Once logged into your admin console, create a new SIP trunk for VoIP.ms by navigating to the '''<tt>Extension</tt>''' page and in the '''<tt>Trunk</tt>''' section select '''<tt>VoIP trunks</tt>''' from the left panel and making the following selections: | ||
| + | * '''Type:''' Register SIP trunk | ||
| + | * '''Provider Name:''' Enter a convenient name, such as "VoIP.ms" or "Main Trunk" | ||
| + | * '''Host Name:''' Enter your preferred POP server to connect to, i.e., <code style="border: thin solid #ccc; padding: 1px 4px; white-space: collapse nowrap;">toronto5.voip.ms</code> | ||
| + | * '''Keep trunk CID:''' Enable if you want the trunk to send its Caller ID for all outgoing calls or disable it if your extensions are going to send their own individual Called ID information | ||
| + | * '''Username:''' Enter your six-digit VoIP.ms account number or the full name of a sub-account | ||
| + | * '''Password:''' Enter the password for the account above | ||
| - | + | Once stored, these trunks are displayed with details such as Provider Name, Type, Hostname/IP and Username, plus options for editing and detecting the trunk. | |
| - | + | * To add a new VoIP trunk, simply click on "Add SIP Trunk." | |
| - | + | * For detailed configuration parameters, click on the ''edit'' icon. | |
| - | + | * To set up Direct Outward Dialing (DOD) for the SIP Trunk, click on the corresponding option. | |
| - | + | * To delete a VoIP trunk, click on the delete icon. | |
| - | + | : '''''Note:''' Ensure to select the same VoIP server that corresponds to your DID number. You can verify the server associated with your VoIP number through your VoIP.ms customer portal under '''DID Numbers''' >> '''Manage DIDs''', in the '''POP''' section. It's crucial to match the server chosen here with the one in your portal to ensure incoming calls ring properly.'' | |
| - | + | * Click the '''Save''' button, but do not click '''Apply Changes''' yet. | |
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| - | Click '''Save''' button, do not click '''Apply Changes''' yet. | + | |
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| + | === Additional SIP settings === | ||
Once you've set up your basic SIP trunk, it's time to optimize some settings. Begin by clicking on the edit icon for your trunk. | Once you've set up your basic SIP trunk, it's time to optimize some settings. Begin by clicking on the edit icon for your trunk. | ||
| - | + | ==== SIP headers ==== | |
| - | : To ensure the proper sending of the '''"FROM"''' header, follow these steps: | + | [[File:SIP-basic settings.png|border|right|576px]] |
| + | To ensure the proper sending of the '''"FROM"''' header, follow these steps: | ||
| + | * Set '''<code>fromuser:</code>''' to your main VoIP.ms account number or sub-account name (replacing <code>100000</code>) | ||
| - | : | + | ==== Audio codecs ==== |
| + | Navigate to "Advanced Settings" and under "Codec Preference," prioritize the codecs supported by VoIP.ms in the order listed: | ||
| + | * '''G.722''' | ||
| + | * '''PCMU''' | ||
| + | * '''G.729''' | ||
| - | + | ==== Trunk Caller ID ==== | |
| + | If you wish to transmit Called ID name and number information from your trunk, utilize the '''<tt>PAI (P-Asserted-Identity)</tt>''' header according to this format: | ||
| + | : '''<code>"CallerIDName" <CallerIDNumber></code>''' | ||
| - | ''' | + | : '''Note:''' Enabling the PAI header without specifying values will default to the CID priority rules from Grandstream, allowing you to pass the Caller ID information through from extensions. To prevent overriding by your trunk, ensure that no Caller ID Number is set in your VoIP.ms portal. |
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| - | ''' | + | ==== NAT Keep Alive ==== |
| - | + | To prevent your modem from closing local SIP ports, set the following configuration options as indicated: | |
| + | * '''Enable Heartbeat Detection:''' Enabled | ||
| + | * '''Heartbeat Frequency:''' <code>50</code> | ||
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After completing the configuration, click on the "Save" button, and optionally, click "Apply Changes" to enact them. Your trunk status should display as "Registered" on your VoIP.ms dashboard. Keep in mind that calls will not be operational until you establish your outbound and inbound routes. | After completing the configuration, click on the "Save" button, and optionally, click "Apply Changes" to enact them. Your trunk status should display as "Registered" on your VoIP.ms dashboard. Keep in mind that calls will not be operational until you establish your outbound and inbound routes. | ||
| - | == Creating | + | === Creating an outbound route === |
| - | Outbound routes play a crucial role in matching your dialing patterns and directing calls through the appropriate trunk. | + | Outbound routes play a crucial role in matching your dialing patterns and directing calls through the appropriate trunk. VoIP.ms recommends including the following patterns in your outbound route: |
| + | ''Ensure that each of your dial patterns is prefixed by the underscore character (<code>_</code>)'' | ||
| + | * <code>_1NXXXXXXXXX</code> | ||
| + | * <code>_NXXXXXXXXX</code> | ||
| + | * <code>_4XXX</code> | ||
| + | * <code>_00.</code> | ||
| + | * <code>_011.</code> | ||
| + | * <code>_033.</code> | ||
| + | * <code>_044.</code> | ||
| - | : '''VoIP.ms | + | To create your outbound routes, navigate to '''<tt>Extension/Trunk</tt>''' ⇒ '''<tt>Outbound routes</tt>''' from the left panel, then click on "Add". In this section, you only need to complete the following fields: |
| + | * '''Calling Rule Name:''' Select any desired name for this route. | ||
| + | * '''Pattern:''' Define the pattern that callers need to dial; you can utilize the suggested dial patterns provided by VoIP.ms, as mentioned above. | ||
| + | * '''Trunk:''' Choose the VoIP.ms trunk through which your call will be routed. | ||
| + | * '''Privilege Level:''' Determine the privilege level required for extensions to utilize this route. If you follow our suggested dial pattern, we recommend setting it to "International" to enable placing international calls. Note that extensions with lower privilege levels won't have access to this route. Hover over the field name for additional information. | ||
| - | : | + | : '''''Note:''' If you wish to incorporate a dial-out prefix, simply type it after the underscore (<code>_</code>) character in your dial patterns. To remove this prefix during dialing, utilize the "Strip" field, where you can specify the number of digits to strip after the underscore character. For example, if you intend to use <code>9</code> as a dial-out prefix, your pattern should be <code>_9NXXXXXXXXX</code>. To remove the <code>9</code> when dialing out, set the "Strip" field to <code>1</code>; this ensures that only one character (in this case, the number <code>9</code>) will be stripped off.'' |
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| - | + | === Creating an inbound route === | |
| + | [[File:Inbound routes setup.png|border|right|576px]] | ||
| + | With your inbound routes configured, you can efficiently manage all incoming calls from multiple phone numbers using just one trunk. This eliminates the need for multiple trunks, as the inbound routes can effectively receive and direct calls to their respective destinations within your UCM. | ||
| - | :: | + | : '''''Note:''' You should not use more than one VoIP.ms trunk on the same device.'' |
| - | + | To set up inbound routes, navigate to '''<tt>Extension/Trunk</tt>''' ⇒ '''<tt>Inbound Routes</tt>''' from the left panel, and then click on '''<tt>Add</tt>'''. | |
| - | + | : [[File:Inbound routes main.png|frameless|320px]] | |
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| - | : [[File:Inbound routes main.png| | + | |
In this section, complete the following fields: | In this section, complete the following fields: | ||
| - | + | * '''Trunks:''' Select your VoIP.ms trunk to specify where incoming calls should be directed. | |
| - | + | * '''Pattern:''' Enter your VoIP.ms DID phone number exactly as it appears in the "Manage DIDs" section of your customer portal. Ensure it is prefixed by the underscore (<code>_</code>) character. Avoid using wildcards and ensure the full number is used to prevent conflicts in the inbound pattern. | |
| - | + | * '''Default Destination:''' Choose the default destination to which your PBX should route incoming calls; this could include an internal extension, an IVR, a ring group, or other locally managed options within your PBX. | |
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== External links == | == External links == | ||
=== Guides and manuals === | === Guides and manuals === | ||
| - | + | * [https://www.grandstream.com/products/ip-pbxs/ucm-series-ip-pbxs/product/clouducm Official product page] | |
| - | : | + | * [https://documentation.grandstream.com/article-categories/cloudUCM Official documentation listing] |
| - | + | ** [https://documentation.grandstream.com/knowledge-base/clouducm-user-manual/ CloudUCM user manual] | |
| - | + | ** [https://documentation.grandstream.com/knowledge-base/clouducm-endpoint-configuration-guide/ CloudUCM endpoint configuration] | |
| - | + | ** [https://documentation.grandstream.com/article-categories/deployment-clouducm/ CloudUCM deployment scenarios] | |
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| + | [[Category:Grandstream|CloudUCM]] | ||
[[Category:PBXes]] | [[Category:PBXes]] | ||
Latest revision as of 09:21, 8 May 2026
Grandstream CloudUCM is a cloud PBX that provides a scalable business communication and collaboration platform with powerful features and integrations that enable teams to be more productive than ever before.
Contents |
Configuration
Logging in
You can log directly to your instance URL or access it from your Grandstream Device Management Cloud System (GDMS) account.
Creating a trunk
Once logged into your admin console, create a new SIP trunk for VoIP.ms by navigating to the Extension page and in the Trunk section select VoIP trunks from the left panel and making the following selections:
- Type: Register SIP trunk
- Provider Name: Enter a convenient name, such as "VoIP.ms" or "Main Trunk"
- Host Name: Enter your preferred POP server to connect to, i.e.,
toronto5.voip.ms - Keep trunk CID: Enable if you want the trunk to send its Caller ID for all outgoing calls or disable it if your extensions are going to send their own individual Called ID information
- Username: Enter your six-digit VoIP.ms account number or the full name of a sub-account
- Password: Enter the password for the account above
Once stored, these trunks are displayed with details such as Provider Name, Type, Hostname/IP and Username, plus options for editing and detecting the trunk.
- To add a new VoIP trunk, simply click on "Add SIP Trunk."
- For detailed configuration parameters, click on the edit icon.
- To set up Direct Outward Dialing (DOD) for the SIP Trunk, click on the corresponding option.
- To delete a VoIP trunk, click on the delete icon.
- Note: Ensure to select the same VoIP server that corresponds to your DID number. You can verify the server associated with your VoIP number through your VoIP.ms customer portal under DID Numbers >> Manage DIDs, in the POP section. It's crucial to match the server chosen here with the one in your portal to ensure incoming calls ring properly.
- Click the Save button, but do not click Apply Changes yet.
Additional SIP settings
Once you've set up your basic SIP trunk, it's time to optimize some settings. Begin by clicking on the edit icon for your trunk.
SIP headers
To ensure the proper sending of the "FROM" header, follow these steps:
- Set
fromuser:to your main VoIP.ms account number or sub-account name (replacing100000)
Audio codecs
Navigate to "Advanced Settings" and under "Codec Preference," prioritize the codecs supported by VoIP.ms in the order listed:
- G.722
- PCMU
- G.729
Trunk Caller ID
If you wish to transmit Called ID name and number information from your trunk, utilize the PAI (P-Asserted-Identity) header according to this format:
-
"CallerIDName" <CallerIDNumber>
- Note: Enabling the PAI header without specifying values will default to the CID priority rules from Grandstream, allowing you to pass the Caller ID information through from extensions. To prevent overriding by your trunk, ensure that no Caller ID Number is set in your VoIP.ms portal.
NAT Keep Alive
To prevent your modem from closing local SIP ports, set the following configuration options as indicated:
- Enable Heartbeat Detection: Enabled
- Heartbeat Frequency:
50
After completing the configuration, click on the "Save" button, and optionally, click "Apply Changes" to enact them. Your trunk status should display as "Registered" on your VoIP.ms dashboard. Keep in mind that calls will not be operational until you establish your outbound and inbound routes.
Creating an outbound route
Outbound routes play a crucial role in matching your dialing patterns and directing calls through the appropriate trunk. VoIP.ms recommends including the following patterns in your outbound route:
Ensure that each of your dial patterns is prefixed by the underscore character (_)
-
_1NXXXXXXXXX -
_NXXXXXXXXX -
_4XXX -
_00. -
_011. -
_033. -
_044.
To create your outbound routes, navigate to Extension/Trunk ⇒ Outbound routes from the left panel, then click on "Add". In this section, you only need to complete the following fields:
- Calling Rule Name: Select any desired name for this route.
- Pattern: Define the pattern that callers need to dial; you can utilize the suggested dial patterns provided by VoIP.ms, as mentioned above.
- Trunk: Choose the VoIP.ms trunk through which your call will be routed.
- Privilege Level: Determine the privilege level required for extensions to utilize this route. If you follow our suggested dial pattern, we recommend setting it to "International" to enable placing international calls. Note that extensions with lower privilege levels won't have access to this route. Hover over the field name for additional information.
- Note: If you wish to incorporate a dial-out prefix, simply type it after the underscore (
_) character in your dial patterns. To remove this prefix during dialing, utilize the "Strip" field, where you can specify the number of digits to strip after the underscore character. For example, if you intend to use9as a dial-out prefix, your pattern should be_9NXXXXXXXXX. To remove the9when dialing out, set the "Strip" field to1; this ensures that only one character (in this case, the number9) will be stripped off.
Creating an inbound route
With your inbound routes configured, you can efficiently manage all incoming calls from multiple phone numbers using just one trunk. This eliminates the need for multiple trunks, as the inbound routes can effectively receive and direct calls to their respective destinations within your UCM.
- Note: You should not use more than one VoIP.ms trunk on the same device.
To set up inbound routes, navigate to Extension/Trunk ⇒ Inbound Routes from the left panel, and then click on Add.
In this section, complete the following fields:
- Trunks: Select your VoIP.ms trunk to specify where incoming calls should be directed.
- Pattern: Enter your VoIP.ms DID phone number exactly as it appears in the "Manage DIDs" section of your customer portal. Ensure it is prefixed by the underscore (
_) character. Avoid using wildcards and ensure the full number is used to prevent conflicts in the inbound pattern. - Default Destination: Choose the default destination to which your PBX should route incoming calls; this could include an internal extension, an IVR, a ring group, or other locally managed options within your PBX.