VoIP.ms Wiki - User contributions [en]

File:VTScada-VoIP-EditDID.png

2026-09-15T23:11:29Z

Scomtois:

File:VTScada-VoIP-CreateSubAccountMenu.png

2026-09-15T22:55:17Z

Scomtois:

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2026-09-15T22:20:41Z

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2026-09-15T22:19:15Z

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2026-09-15T22:18:08Z

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2026-09-15T22:14:09Z

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2026-09-15T22:03:51Z

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Hello Gubby

2026-08-31T21:08:02Z

Scomtois: /* Your phone rings first, your AI receptionist answers missed calls */

[[File:01-hello-gubby-logo.png|right|500px]]
Hello Gubby is an AI receptionist platform that grows into the AI front desk for your business.

'''Your AI receptionist answers every call in seconds with natural, human-sounding voices.''' It greets callers with your business name or a greeting of your choice, and answers questions about your services. It captures lead details and books appointments. When a caller needs a person, it warm transfers the call to you or your team, with unlimited warm transfers and unlimited notification recipients at no extra cost.

Every call is captured. After each call, you receive a summary, transcript, and recording on your email, in your Hello Gubby account, and in your CRM. Connect GoHighLevel or HubSpot directly from your Hello Gubby account, or sync your calendar for bookings. Webhooks are live, and Zapier is coming soon.

Hello Gubby is a no-code platform designed for busy business owners who need a done-for-you (DFY) setup at do-it-yourself (DIY) prices. Train your AI receptionist from your website or upload company materials, and go live in less than 5 minutes. When you are ready, add your AI receptionist to talk to your website visitors over voice or chat, and upload your own video or an AI receptionist video to give life to your website.

For more details or to sign up, visit: [https://hellogubby.ai hellogubby.ai]

__TOC__

== Before you start ==

To connect a number to Hello Gubby you will need:

# '''A VoIP.ms account with an active DID (phone number).''' If you don't have a number yet, follow: [[Order a DID Number]]
#:* You can leave the routing settings at their default values during the purchase. You will configure routing based on your use case (i.e., all calls answered by your AI receptionist, missed calls only, etc.) later in this guide.
# '''The DID on the pay as you go plan.'''
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Plan type</strong><br>Flat rate plans are limited to 2 simultaneous incoming calls, which limits how many callers your AI receptionist can answer at the same time. The pay as you go plan has no such restriction and is the recommended plan for AI receptionists. To change your plan, see the [[Changing DID Billing Plan]] article on this wiki.</div>
# '''A Hello Gubby account on a paid plan.''' Sign up at [https://hellogubby.ai hellogubby.ai]. Each plan comes with a 7-day free trial so try before you pay.

== Hello Gubby configuration ==

After you sign up, you can explore your Hello Gubby portal and test your AI receptionist before paying. You can set up your receptionist, modify its settings, and make a web call to listen to it, all without putting in your credit card. See this video walkthrough of the sign up and setup: [https://youtu.be/w0E3PMtBRIA?si=6CgC3OCARCjVKnys&t=360 How to sign up for Hello Gubby]

Once you pay, your subscription is active and you are navigated from Stripe back to your Hello Gubby portal, which takes you to your business number setup. You will see the You're live! confirmation, as shown below.

=== Connecting your number ===

# On the '''You're live!''' Confirmation screen, click on '''Continue'''.
#: [[File:hello-gubby-02-youre-live-continue.png|none|700px]]
# Choose how you want your AI front desk to answer calls: select '''Bring your own number''', then click on '''Continue'''.
#: [[File:hello-gubby-03-bring-your-own-number.png|none|700px]]
# You are taken to the '''SIP Trunk''' section of your settings. Click on the '''VoIP.ms''' card.
#: [[File:hello-gubby-04-sip-trunk-voipms-card.png|none|700px]]
#:If you skipped this step after payment, you can reach it anytime from '''Settings >> SIP Trunk''' in your Hello Gubby dashboard.
# '''Your VoIP.ms phone number:''' Enter your VoIP.ms DID with your country code (E.g. +1 778 819 7943).
# Click on '''Connect'''.
#: [[File:hello-gubby-05-connect-voipms-number.png|none|700px]]

=== Copy your SIP address ===

Once your number is connected, Hello Gubby shows it as your AI front desk line and displays your SIP address, along with the steps to complete in VoIP.ms.

# Click on '''Copy''' next to the SIP address.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Copy the address exactly</strong><br>Paste the address in VoIP.ms exactly as shown, with your number at the front. A shortened address without the number still connects calls, but your AI receptionist cannot tell which number was dialed and will answer with a generic greeting instead of your business greeting.</div>
#:[[File:hello-gubby-06-copy-sip-address.png|none|700px]]

The next section walks you through the two steps in your VoIP.ms portal.

== VoIP.ms configuration ==

You will now use the SIP address you copied from Hello Gubby in two places in your VoIP.ms portal: first to create a SIP URI, then to route your number to it.

=== Create the SIP URI ===

# In your VoIP.ms portal, go to '''DID Numbers >> SIP URI's'''.
#:[[File:hello-gubby-07-voipms-sip-uri-menu.png|none|700px]]
# Click on '''Add SIP uri'''.
#:[[File:hello-gubby-08-add-sip-uri.png|none|700px]]
# '''SIP URI:''' Paste the address you copied from your Hello Gubby dashboard.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 The sip: prefix is already there</strong><br>The field starts with sip: on its left. Paste only the address as copied (E.g. 17788197943@sip.hellogubby.ai). Do not type sip: again.</div>
# '''Description:''' Any name of your choice (E.g. Hello Gubby AI Receptionist).
# '''CallerID Override:''' Select '''I use a system capable of passing its own CallerID'''.
# '''CallerID E164:''' Leave as '''Disabled'''.
# Click on '''Save'''.
#:[[File:hello-gubby-09-create-sip-uri-form.png|none|700px]]

Your SIP URI is created and appears in the list.

: [[File:10-sip-uri-created.png|none|700px]]

=== Open your DID settings ===

# Go to '''DID Numbers >> Manage DID(s)'''.
#:[[File:hello-gubby-11-manage-dids-menu.png|none|700px]]
# Find your VoIP.ms number and click on the '''Edit DID''' (pencil) icon.
#:[[File:hello-gubby-12-edit-did.png|none|700px]]

=== Update the DID routing ===

# Under '''Routing Settings''', select '''SIP URI'''.
# In the dropdown next to it, choose the SIP URI you created.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Select the radio button, not just the dropdown</strong><br>Your SIP URI appearing in the dropdown is not enough. The round radio button next to SIP URI must be selected, or calls will keep going to the previous destination.</div>
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 This option routes ALL calls to your AI receptionist</strong><br>With SIP URI selected as your Main routing, every call to this number is answered by your AI receptionist. If you want your phone to ring first and your AI receptionist to answer only missed calls, or other setups, see '''Choosing when your AI receptionist answers''' below.</div>
#: [[File:hello-gubby-13-route-did-to-sip-uri.png|none|700px]]
# Scroll down and click on '''Apply Changes'''.

That's it! Every call to your VoIP.ms number is now answered by your AI receptionist. To have your AI receptionist answer only some calls instead, continue to the routing options below.

== Test your number ==

Call your VoIP.ms number from any phone.

Your AI receptionist should answer within a few seconds with your business greeting, for example: "Thanks for calling Bright Nest Cleaning, this is Filzah. How can I help you today?"

Then check your Hello Gubby dashboard: the call appears in your '''Call Log Inbox''' within a minute, with the summary, transcript, and recording.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 A generic greeting means the address was shortened</strong><br>If your AI receptionist answers with a generic Hello Gubby greeting instead of your business greeting, the SIP URI in VoIP.ms is missing your number at the front. Edit it under '''DID Numbers >> SIP URI's''' so it exactly matches the address shown in your Hello Gubby dashboard, with your number in front.</div>

== Choosing when your AI receptionist answers ==

The setup above routes every call to your AI receptionist. VoIP.ms routing gives you other options depending on how you want to run your business. All of them are configured on the same '''Edit DID Settings''' page, and none of them require any change in your Hello Gubby account.

=== Your AI receptionist answers every call ===

The default setup from this guide: '''Main routing = SIP URI'''. Your AI receptionist answers every call in seconds, 24/7, and warm transfers callers to you or your team when needed. Best for businesses that miss calls during jobs, after hours, or at busy times, and want every call captured.

=== Your phone rings first, your AI receptionist answers missed calls ===

Your phone rings normally, and your AI receptionist only picks up the calls you can't take. No call ever goes to voicemail again.

On your DID's '''Routing Settings''', the tabs across the top control where calls go in each situation:

'''4. Main:''' set to where your phone is. Select '''SIP/IAX''' if you use a softphone registered to your VoIP.ms account, or '''Call Forwarding''' with your mobile number (create the forwarding entry first under '''DID Numbers >> Call Forwarding''').

: [[File:hello-gubby-14-main-routing-phone.png|none|700px]]

'''5. If Busy:''' select '''SIP URI''' and choose the one you created.

'''6. If Unreachable:''' select '''SIP URI''' and choose the one you created.

'''7. If No Answer:''' select '''SIP URI''' and choose the one you created.

: [[File:hello-gubby-15-failover-routing-sip-uri.png|none|700px]]

'''8.''' Scroll down to '''Ring Time in seconds''' and set it to '''20 s (~4 Rings)'''. This is how long your phone rings before your AI receptionist takes the call.

'''9.''' Click on '''Apply Changes'''.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Watch out for your phone's own voicemail</strong><br>If your Main destination is call forwarding to a mobile phone, the mobile carrier's voicemail competes with this setup. If your carrier voicemail answers at 18 seconds and your VoIP.ms Ring Time is 20, voicemail takes every missed call before your AI receptionist can. Set your Ring Time shorter than your phone's voicemail delay, or disable your carrier voicemail. Softphone setups do not have this issue.</div>

Now your phone rings first on every call. If you don't answer within the ring time, are on another call, or your phone is offline, your AI receptionist answers and captures the lead. For more details on forwarding entries, see: [[Call Forwarding]]

=== Business hours to your phone, after hours to your AI receptionist ===

Your team answers during the day, and your AI receptionist covers evenings, weekends, and holidays.

:1. Go to '''DID Numbers >> Time Conditions''' and click on '''Create a new Time Condition'''.
:2. Set the time and day criteria for your business hours.
:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 VoIP.ms servers run on Eastern Time</strong><br>All VoIP.ms servers are set to Eastern Time, regardless of your location. Enter your hours converted to ET. For example, a Vancouver business open 9:00 to 17:00 Pacific enters 12:00 to 20:00. Compare your local time with the Current server time shown on the page.</div>
: [[File:hello-gubby-16-time-conditions-hours.png|none|700px]]
:1. '''Destination if one of the conditions matches server time (during business hours):''' select where your phone is, for example '''SIP/IAX''' or '''Call Forwarding'''.
:2. '''Destination if no conditions matches server time (after hours):''' select '''SIP URI''' and choose the one you created.
:3. '''Name of Time Condition:''' Any name of your choice (E.g. Business Hours). Click on '''Save Time Condition'''.
: [[File:hello-gubby-17-time-conditions-destinations.png|none|700px]]
:4. Go to '''DID Numbers >> Manage DID(s)''', edit your number, and under '''Routing Settings''' select '''Time Conditions''' with the condition you created. Click on '''Apply Changes'''.
: [[File:hello-gubby-18-select-time-condition-on-did.png|none|700px]]

Statutory holidays are not automatic: a time condition is a weekly schedule, so on holidays your phone still receives calls during the configured hours unless you temporarily adjust the condition. For more details, see: [[Time Conditions]]

=== Keep your existing main line and forward it to your AI receptionist ===

If your business number is with another provider and you don't want to move it, you can still use your AI receptionist. Set up call forwarding on your existing line (from your current provider's settings) to your VoIP.ms number, either for all calls or only when you don't answer. Your VoIP.ms number then routes to your AI receptionist as configured in this guide. Your published number never changes.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Check the caller number after your first forwarded call</strong><br>Some carriers replace the caller's number with the forwarding number when they forward a call. Make one test call and check the caller number shown in your Hello Gubby call log. Also note that forwarded calls are billed by both providers: your carrier's forwarding rate plus VoIP.ms per-minute usage.</div>

== Outbound calls and transfers ==

Warm transfers to your team and outbound calls from your VoIP.ms number: '''Coming soon!'''

== Useful resources ==

* '''Hello Gubby:''' [https://hellogubby.ai hellogubby.ai]
* '''Hello Gubby sign up and setup video:''' [https://youtu.be/w0E3PMtBRIA?t=360 YouTube walkthrough]
* '''VoIP.ms SIP URI feature:''' [[SIP URI]]
* '''VoIP.ms DID ordering:''' [[Order a DID Number]]
* '''VoIP.ms DID plans:''' [[Changing DID Billing Plan]]
* '''VoIP.ms Time Conditions:''' [[Time Conditions]]
* '''VoIP.ms Call Forwarding:''' [[Call Forwarding]]

Hello Gubby

2026-08-31T21:07:34Z

Scomtois: /* Your phone rings first, your AI receptionist answers missed calls */

[[File:01-hello-gubby-logo.png|right|500px]]
Hello Gubby is an AI receptionist platform that grows into the AI front desk for your business.

'''Your AI receptionist answers every call in seconds with natural, human-sounding voices.''' It greets callers with your business name or a greeting of your choice, and answers questions about your services. It captures lead details and books appointments. When a caller needs a person, it warm transfers the call to you or your team, with unlimited warm transfers and unlimited notification recipients at no extra cost.

Every call is captured. After each call, you receive a summary, transcript, and recording on your email, in your Hello Gubby account, and in your CRM. Connect GoHighLevel or HubSpot directly from your Hello Gubby account, or sync your calendar for bookings. Webhooks are live, and Zapier is coming soon.

Hello Gubby is a no-code platform designed for busy business owners who need a done-for-you (DFY) setup at do-it-yourself (DIY) prices. Train your AI receptionist from your website or upload company materials, and go live in less than 5 minutes. When you are ready, add your AI receptionist to talk to your website visitors over voice or chat, and upload your own video or an AI receptionist video to give life to your website.

For more details or to sign up, visit: [https://hellogubby.ai hellogubby.ai]

__TOC__

== Before you start ==

To connect a number to Hello Gubby you will need:

# '''A VoIP.ms account with an active DID (phone number).''' If you don't have a number yet, follow: [[Order a DID Number]]
#:* You can leave the routing settings at their default values during the purchase. You will configure routing based on your use case (i.e., all calls answered by your AI receptionist, missed calls only, etc.) later in this guide.
# '''The DID on the pay as you go plan.'''
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Plan type</strong><br>Flat rate plans are limited to 2 simultaneous incoming calls, which limits how many callers your AI receptionist can answer at the same time. The pay as you go plan has no such restriction and is the recommended plan for AI receptionists. To change your plan, see the [[Changing DID Billing Plan]] article on this wiki.</div>
# '''A Hello Gubby account on a paid plan.''' Sign up at [https://hellogubby.ai hellogubby.ai]. Each plan comes with a 7-day free trial so try before you pay.

== Hello Gubby configuration ==

After you sign up, you can explore your Hello Gubby portal and test your AI receptionist before paying. You can set up your receptionist, modify its settings, and make a web call to listen to it, all without putting in your credit card. See this video walkthrough of the sign up and setup: [https://youtu.be/w0E3PMtBRIA?si=6CgC3OCARCjVKnys&t=360 How to sign up for Hello Gubby]

Once you pay, your subscription is active and you are navigated from Stripe back to your Hello Gubby portal, which takes you to your business number setup. You will see the You're live! confirmation, as shown below.

=== Connecting your number ===

# On the '''You're live!''' Confirmation screen, click on '''Continue'''.
#: [[File:hello-gubby-02-youre-live-continue.png|none|700px]]
# Choose how you want your AI front desk to answer calls: select '''Bring your own number''', then click on '''Continue'''.
#: [[File:hello-gubby-03-bring-your-own-number.png|none|700px]]
# You are taken to the '''SIP Trunk''' section of your settings. Click on the '''VoIP.ms''' card.
#: [[File:hello-gubby-04-sip-trunk-voipms-card.png|none|700px]]
#:If you skipped this step after payment, you can reach it anytime from '''Settings >> SIP Trunk''' in your Hello Gubby dashboard.
# '''Your VoIP.ms phone number:''' Enter your VoIP.ms DID with your country code (E.g. +1 778 819 7943).
# Click on '''Connect'''.
#: [[File:hello-gubby-05-connect-voipms-number.png|none|700px]]

=== Copy your SIP address ===

Once your number is connected, Hello Gubby shows it as your AI front desk line and displays your SIP address, along with the steps to complete in VoIP.ms.

# Click on '''Copy''' next to the SIP address.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Copy the address exactly</strong><br>Paste the address in VoIP.ms exactly as shown, with your number at the front. A shortened address without the number still connects calls, but your AI receptionist cannot tell which number was dialed and will answer with a generic greeting instead of your business greeting.</div>
#:[[File:hello-gubby-06-copy-sip-address.png|none|700px]]

The next section walks you through the two steps in your VoIP.ms portal.

== VoIP.ms configuration ==

You will now use the SIP address you copied from Hello Gubby in two places in your VoIP.ms portal: first to create a SIP URI, then to route your number to it.

=== Create the SIP URI ===

# In your VoIP.ms portal, go to '''DID Numbers >> SIP URI's'''.
#:[[File:hello-gubby-07-voipms-sip-uri-menu.png|none|700px]]
# Click on '''Add SIP uri'''.
#:[[File:hello-gubby-08-add-sip-uri.png|none|700px]]
# '''SIP URI:''' Paste the address you copied from your Hello Gubby dashboard.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 The sip: prefix is already there</strong><br>The field starts with sip: on its left. Paste only the address as copied (E.g. 17788197943@sip.hellogubby.ai). Do not type sip: again.</div>
# '''Description:''' Any name of your choice (E.g. Hello Gubby AI Receptionist).
# '''CallerID Override:''' Select '''I use a system capable of passing its own CallerID'''.
# '''CallerID E164:''' Leave as '''Disabled'''.
# Click on '''Save'''.
#:[[File:hello-gubby-09-create-sip-uri-form.png|none|700px]]

Your SIP URI is created and appears in the list.

: [[File:10-sip-uri-created.png|none|700px]]

=== Open your DID settings ===

# Go to '''DID Numbers >> Manage DID(s)'''.
#:[[File:hello-gubby-11-manage-dids-menu.png|none|700px]]
# Find your VoIP.ms number and click on the '''Edit DID''' (pencil) icon.
#:[[File:hello-gubby-12-edit-did.png|none|700px]]

=== Update the DID routing ===

# Under '''Routing Settings''', select '''SIP URI'''.
# In the dropdown next to it, choose the SIP URI you created.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Select the radio button, not just the dropdown</strong><br>Your SIP URI appearing in the dropdown is not enough. The round radio button next to SIP URI must be selected, or calls will keep going to the previous destination.</div>
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 This option routes ALL calls to your AI receptionist</strong><br>With SIP URI selected as your Main routing, every call to this number is answered by your AI receptionist. If you want your phone to ring first and your AI receptionist to answer only missed calls, or other setups, see '''Choosing when your AI receptionist answers''' below.</div>
#: [[File:hello-gubby-13-route-did-to-sip-uri.png|none|700px]]
# Scroll down and click on '''Apply Changes'''.

That's it! Every call to your VoIP.ms number is now answered by your AI receptionist. To have your AI receptionist answer only some calls instead, continue to the routing options below.

== Test your number ==

Call your VoIP.ms number from any phone.

Your AI receptionist should answer within a few seconds with your business greeting, for example: "Thanks for calling Bright Nest Cleaning, this is Filzah. How can I help you today?"

Then check your Hello Gubby dashboard: the call appears in your '''Call Log Inbox''' within a minute, with the summary, transcript, and recording.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 A generic greeting means the address was shortened</strong><br>If your AI receptionist answers with a generic Hello Gubby greeting instead of your business greeting, the SIP URI in VoIP.ms is missing your number at the front. Edit it under '''DID Numbers >> SIP URI's''' so it exactly matches the address shown in your Hello Gubby dashboard, with your number in front.</div>

== Choosing when your AI receptionist answers ==

The setup above routes every call to your AI receptionist. VoIP.ms routing gives you other options depending on how you want to run your business. All of them are configured on the same '''Edit DID Settings''' page, and none of them require any change in your Hello Gubby account.

=== Your AI receptionist answers every call ===

The default setup from this guide: '''Main routing = SIP URI'''. Your AI receptionist answers every call in seconds, 24/7, and warm transfers callers to you or your team when needed. Best for businesses that miss calls during jobs, after hours, or at busy times, and want every call captured.

=== Your phone rings first, your AI receptionist answers missed calls ===

Your phone rings normally, and your AI receptionist only picks up the calls you can't take. No call ever goes to voicemail again.

On your DID's '''Routing Settings''', the tabs across the top control where calls go in each situation:

'''1. Main:''' set to where your phone is. Select '''SIP/IAX''' if you use a softphone registered to your VoIP.ms account, or '''Call Forwarding''' with your mobile number (create the forwarding entry first under '''DID Numbers >> Call Forwarding''').

: [[File:hello-gubby-14-main-routing-phone.png|none|700px]]

'''2. If Busy:''' select '''SIP URI''' and choose the one you created.

'''3. If Unreachable:''' select '''SIP URI''' and choose the one you created.

'''4. If No Answer:''' select '''SIP URI''' and choose the one you created.

: [[File:hello-gubby-15-failover-routing-sip-uri.png|none|700px]]

'''5.''' Scroll down to '''Ring Time in seconds''' and set it to '''20 s (~4 Rings)'''. This is how long your phone rings before your AI receptionist takes the call.

'''6.''' Click on '''Apply Changes'''.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Watch out for your phone's own voicemail</strong><br>If your Main destination is call forwarding to a mobile phone, the mobile carrier's voicemail competes with this setup. If your carrier voicemail answers at 18 seconds and your VoIP.ms Ring Time is 20, voicemail takes every missed call before your AI receptionist can. Set your Ring Time shorter than your phone's voicemail delay, or disable your carrier voicemail. Softphone setups do not have this issue.</div>

Now your phone rings first on every call. If you don't answer within the ring time, are on another call, or your phone is offline, your AI receptionist answers and captures the lead. For more details on forwarding entries, see: [[Call Forwarding]]

=== Business hours to your phone, after hours to your AI receptionist ===

Your team answers during the day, and your AI receptionist covers evenings, weekends, and holidays.

:1. Go to '''DID Numbers >> Time Conditions''' and click on '''Create a new Time Condition'''.
:2. Set the time and day criteria for your business hours.
:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 VoIP.ms servers run on Eastern Time</strong><br>All VoIP.ms servers are set to Eastern Time, regardless of your location. Enter your hours converted to ET. For example, a Vancouver business open 9:00 to 17:00 Pacific enters 12:00 to 20:00. Compare your local time with the Current server time shown on the page.</div>
: [[File:hello-gubby-16-time-conditions-hours.png|none|700px]]
:1. '''Destination if one of the conditions matches server time (during business hours):''' select where your phone is, for example '''SIP/IAX''' or '''Call Forwarding'''.
:2. '''Destination if no conditions matches server time (after hours):''' select '''SIP URI''' and choose the one you created.
:3. '''Name of Time Condition:''' Any name of your choice (E.g. Business Hours). Click on '''Save Time Condition'''.
: [[File:hello-gubby-17-time-conditions-destinations.png|none|700px]]
:4. Go to '''DID Numbers >> Manage DID(s)''', edit your number, and under '''Routing Settings''' select '''Time Conditions''' with the condition you created. Click on '''Apply Changes'''.
: [[File:hello-gubby-18-select-time-condition-on-did.png|none|700px]]

Statutory holidays are not automatic: a time condition is a weekly schedule, so on holidays your phone still receives calls during the configured hours unless you temporarily adjust the condition. For more details, see: [[Time Conditions]]

=== Keep your existing main line and forward it to your AI receptionist ===

If your business number is with another provider and you don't want to move it, you can still use your AI receptionist. Set up call forwarding on your existing line (from your current provider's settings) to your VoIP.ms number, either for all calls or only when you don't answer. Your VoIP.ms number then routes to your AI receptionist as configured in this guide. Your published number never changes.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Check the caller number after your first forwarded call</strong><br>Some carriers replace the caller's number with the forwarding number when they forward a call. Make one test call and check the caller number shown in your Hello Gubby call log. Also note that forwarded calls are billed by both providers: your carrier's forwarding rate plus VoIP.ms per-minute usage.</div>

== Outbound calls and transfers ==

Warm transfers to your team and outbound calls from your VoIP.ms number: '''Coming soon!'''

== Useful resources ==

* '''Hello Gubby:''' [https://hellogubby.ai hellogubby.ai]
* '''Hello Gubby sign up and setup video:''' [https://youtu.be/w0E3PMtBRIA?t=360 YouTube walkthrough]
* '''VoIP.ms SIP URI feature:''' [[SIP URI]]
* '''VoIP.ms DID ordering:''' [[Order a DID Number]]
* '''VoIP.ms DID plans:''' [[Changing DID Billing Plan]]
* '''VoIP.ms Time Conditions:''' [[Time Conditions]]
* '''VoIP.ms Call Forwarding:''' [[Call Forwarding]]

Hello Gubby

2026-08-31T21:06:39Z

Scomtois: /* Business hours to your phone, after hours to your AI receptionist */

Hello Gubby

2026-08-28T19:02:04Z

Scomtois: /* Business hours to your phone, after hours to your AI receptionist */

[[File:01-hello-gubby-logo.png|right|500px]]
Hello Gubby is an AI receptionist platform that grows into the AI front desk for your business.

'''Your AI receptionist answers every call in seconds with natural, human-sounding voices.''' It greets callers with your business name or a greeting of your choice, and answers questions about your services. It captures lead details and books appointments. When a caller needs a person, it warm transfers the call to you or your team, with unlimited warm transfers and unlimited notification recipients at no extra cost.

Every call is captured. After each call, you receive a summary, transcript, and recording on your email, in your Hello Gubby account, and in your CRM. Connect GoHighLevel or HubSpot directly from your Hello Gubby account, or sync your calendar for bookings. Webhooks are live, and Zapier is coming soon.

Hello Gubby is a no-code platform designed for busy business owners who need a done-for-you (DFY) setup at do-it-yourself (DIY) prices. Train your AI receptionist from your website or upload company materials, and go live in less than 5 minutes. When you are ready, add your AI receptionist to talk to your website visitors over voice or chat, and upload your own video or an AI receptionist video to give life to your website.

For more details or to sign up, visit: [https://hellogubby.ai hellogubby.ai]

__TOC__

== Before you start ==

To connect a number to Hello Gubby you will need:

# '''A VoIP.ms account with an active DID (phone number).''' If you don't have a number yet, follow: [[Order a DID Number]]
#:* You can leave the routing settings at their default values during the purchase. You will configure routing based on your use case (i.e., all calls answered by your AI receptionist, missed calls only, etc.) later in this guide.
# '''The DID on the pay as you go plan.'''
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Plan type</strong><br>Flat rate plans are limited to 2 simultaneous incoming calls, which limits how many callers your AI receptionist can answer at the same time. The pay as you go plan has no such restriction and is the recommended plan for AI receptionists. To change your plan, see the [[Changing DID Billing Plan]] article on this wiki.</div>
# '''A Hello Gubby account on a paid plan.''' Sign up at [https://hellogubby.ai hellogubby.ai]. Each plan comes with a 7-day free trial so try before you pay.

== Hello Gubby configuration ==

After you sign up, you can explore your Hello Gubby portal and test your AI receptionist before paying. You can set up your receptionist, modify its settings, and make a web call to listen to it, all without putting in your credit card. See this video walkthrough of the sign up and setup: [https://youtu.be/w0E3PMtBRIA?si=6CgC3OCARCjVKnys&t=360 How to sign up for Hello Gubby]

Once you pay, your subscription is active and you are navigated from Stripe back to your Hello Gubby portal, which takes you to your business number setup. You will see the You're live! confirmation, as shown below.

=== Connecting your number ===

# On the '''You're live!''' Confirmation screen, click on '''Continue'''.
#: [[File:hello-gubby-02-youre-live-continue.png|none|700px]]
# Choose how you want your AI front desk to answer calls: select '''Bring your own number''', then click on '''Continue'''.
#: [[File:hello-gubby-03-bring-your-own-number.png|none|700px]]
# You are taken to the '''SIP Trunk''' section of your settings. Click on the '''VoIP.ms''' card.
#: [[File:hello-gubby-04-sip-trunk-voipms-card.png|none|700px]]
#:If you skipped this step after payment, you can reach it anytime from '''Settings >> SIP Trunk''' in your Hello Gubby dashboard.
# '''Your VoIP.ms phone number:''' Enter your VoIP.ms DID with your country code (E.g. +1 778 819 7943).
# Click on '''Connect'''.
#: [[File:hello-gubby-05-connect-voipms-number.png|none|700px]]

=== Copy your SIP address ===

Once your number is connected, Hello Gubby shows it as your AI front desk line and displays your SIP address, along with the steps to complete in VoIP.ms.

# Click on '''Copy''' next to the SIP address.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Copy the address exactly</strong><br>Paste the address in VoIP.ms exactly as shown, with your number at the front. A shortened address without the number still connects calls, but your AI receptionist cannot tell which number was dialed and will answer with a generic greeting instead of your business greeting.</div>
#:[[File:hello-gubby-06-copy-sip-address.png|none|700px]]

The next section walks you through the two steps in your VoIP.ms portal.

== VoIP.ms configuration ==

You will now use the SIP address you copied from Hello Gubby in two places in your VoIP.ms portal: first to create a SIP URI, then to route your number to it.

=== Create the SIP URI ===

# In your VoIP.ms portal, go to '''DID Numbers >> SIP URI's'''.
#:[[File:hello-gubby-07-voipms-sip-uri-menu.png|none|700px]]
# Click on '''Add SIP uri'''.
#:[[File:hello-gubby-08-add-sip-uri.png|none|700px]]
# '''SIP URI:''' Paste the address you copied from your Hello Gubby dashboard.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 The sip: prefix is already there</strong><br>The field starts with sip: on its left. Paste only the address as copied (E.g. 17788197943@sip.hellogubby.ai). Do not type sip: again.</div>
# '''Description:''' Any name of your choice (E.g. Hello Gubby AI Receptionist).
# '''CallerID Override:''' Select '''I use a system capable of passing its own CallerID'''.
# '''CallerID E164:''' Leave as '''Disabled'''.
# Click on '''Save'''.
#:[[File:hello-gubby-09-create-sip-uri-form.png|none|700px]]

Your SIP URI is created and appears in the list.

: [[File:10-sip-uri-created.png|none|700px]]

=== Open your DID settings ===

# Go to '''DID Numbers >> Manage DID(s)'''.
#:[[File:hello-gubby-11-manage-dids-menu.png|none|700px]]
# Find your VoIP.ms number and click on the '''Edit DID''' (pencil) icon.
#:[[File:hello-gubby-12-edit-did.png|none|700px]]

=== Update the DID routing ===

# Under '''Routing Settings''', select '''SIP URI'''.
# In the dropdown next to it, choose the SIP URI you created.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Select the radio button, not just the dropdown</strong><br>Your SIP URI appearing in the dropdown is not enough. The round radio button next to SIP URI must be selected, or calls will keep going to the previous destination.</div>
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 This option routes ALL calls to your AI receptionist</strong><br>With SIP URI selected as your Main routing, every call to this number is answered by your AI receptionist. If you want your phone to ring first and your AI receptionist to answer only missed calls, or other setups, see '''Choosing when your AI receptionist answers''' below.</div>
#: [[File:hello-gubby-13-route-did-to-sip-uri.png|none|700px]]
# Scroll down and click on '''Apply Changes'''.

That's it! Every call to your VoIP.ms number is now answered by your AI receptionist. To have your AI receptionist answer only some calls instead, continue to the routing options below.

== Test your number ==

Call your VoIP.ms number from any phone.

Your AI receptionist should answer within a few seconds with your business greeting, for example: "Thanks for calling Bright Nest Cleaning, this is Filzah. How can I help you today?"

Then check your Hello Gubby dashboard: the call appears in your '''Call Log Inbox''' within a minute, with the summary, transcript, and recording.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 A generic greeting means the address was shortened</strong><br>If your AI receptionist answers with a generic Hello Gubby greeting instead of your business greeting, the SIP URI in VoIP.ms is missing your number at the front. Edit it under '''DID Numbers >> SIP URI's''' so it exactly matches the address shown in your Hello Gubby dashboard, with your number in front.</div>

== Choosing when your AI receptionist answers ==

The setup above routes every call to your AI receptionist. VoIP.ms routing gives you other options depending on how you want to run your business. All of them are configured on the same '''Edit DID Settings''' page, and none of them require any change in your Hello Gubby account.

=== Your AI receptionist answers every call ===

The default setup from this guide: '''Main routing = SIP URI'''. Your AI receptionist answers every call in seconds, 24/7, and warm transfers callers to you or your team when needed. Best for businesses that miss calls during jobs, after hours, or at busy times, and want every call captured.

=== Your phone rings first, your AI receptionist answers missed calls ===

Your phone rings normally, and your AI receptionist only picks up the calls you can't take. No call ever goes to voicemail again.

On your DID's '''Routing Settings''', the tabs across the top control where calls go in each situation:

'''4. Main:''' set to where your phone is. Select '''SIP/IAX''' if you use a softphone registered to your VoIP.ms account, or '''Call Forwarding''' with your mobile number (create the forwarding entry first under '''DID Numbers >> Call Forwarding''').

: [[File:hello-gubby-14-main-routing-phone.png|none|700px]]

'''5. If Busy:''' select '''SIP URI''' and choose the one you created.

'''6. If Unreachable:''' select '''SIP URI''' and choose the one you created.

'''7. If No Answer:''' select '''SIP URI''' and choose the one you created.

: [[File:hello-gubby-15-failover-routing-sip-uri.png|none|700px]]

'''8.''' Scroll down to '''Ring Time in seconds''' and set it to '''20 s (~4 Rings)'''. This is how long your phone rings before your AI receptionist takes the call.

'''9.''' Click on '''Apply Changes'''.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Watch out for your phone's own voicemail</strong><br>If your Main destination is call forwarding to a mobile phone, the mobile carrier's voicemail competes with this setup. If your carrier voicemail answers at 18 seconds and your VoIP.ms Ring Time is 20, voicemail takes every missed call before your AI receptionist can. Set your Ring Time shorter than your phone's voicemail delay, or disable your carrier voicemail. Softphone setups do not have this issue.</div>

Now your phone rings first on every call. If you don't answer within the ring time, are on another call, or your phone is offline, your AI receptionist answers and captures the lead. For more details on forwarding entries, see: [[Call Forwarding]]

=== Business hours to your phone, after hours to your AI receptionist ===

Your team answers during the day, and your AI receptionist covers evenings, weekends, and holidays.

:1. Go to '''DID Numbers >> Time Conditions''' and click on '''Create a new Time Condition'''.
:2. Set the time and day criteria for your business hours.
:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 VoIP.ms servers run on Eastern Time</strong><br>All VoIP.ms servers are set to Eastern Time, regardless of your location. Enter your hours converted to ET. For example, a Vancouver business open 9:00 to 17:00 Pacific enters 12:00 to 20:00. Compare your local time with the Current server time shown on the page.</div>
: [[File:hello-gubby-16-time-conditions-hours.png|none|700px]]
:3. '''Destination if one of the conditions matches server time (during business hours):''' select where your phone is, for example '''SIP/IAX''' or '''Call Forwarding'''.
:4. '''Destination if no conditions matches server time (after hours):''' select '''SIP URI''' and choose the one you created.
:5. '''Name of Time Condition:''' Any name of your choice (E.g. Business Hours). Click on '''Save Time Condition'''.
: [[File:hello-gubby-17-time-conditions-destinations.png|none|700px]]
:6. Go to '''DID Numbers >> Manage DID(s)''', edit your number, and under '''Routing Settings''' select '''Time Conditions''' with the condition you created. Click on '''Apply Changes'''.
: [[File:hello-gubby-18-select-time-condition-on-did.png|none|700px]]

Statutory holidays are not automatic: a time condition is a weekly schedule, so on holidays your phone still receives calls during the configured hours unless you temporarily adjust the condition. For more details, see: [[Time Conditions]]

=== Keep your existing main line and forward it to your AI receptionist ===

If your business number is with another provider and you don't want to move it, you can still use your AI receptionist. Set up call forwarding on your existing line (from your current provider's settings) to your VoIP.ms number, either for all calls or only when you don't answer. Your VoIP.ms number then routes to your AI receptionist as configured in this guide. Your published number never changes.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Check the caller number after your first forwarded call</strong><br>Some carriers replace the caller's number with the forwarding number when they forward a call. Make one test call and check the caller number shown in your Hello Gubby call log. Also note that forwarded calls are billed by both providers: your carrier's forwarding rate plus VoIP.ms per-minute usage.</div>

== Outbound calls and transfers ==

Warm transfers to your team and outbound calls from your VoIP.ms number: '''Coming soon!'''

== Useful resources ==

* '''Hello Gubby:''' [https://hellogubby.ai hellogubby.ai]
* '''Hello Gubby sign up and setup video:''' [https://youtu.be/w0E3PMtBRIA?t=360 YouTube walkthrough]
* '''VoIP.ms SIP URI feature:''' [[SIP URI]]
* '''VoIP.ms DID ordering:''' [[Order a DID Number]]
* '''VoIP.ms DID plans:''' [[Changing DID Billing Plan]]
* '''VoIP.ms Time Conditions:''' [[Time Conditions]]
* '''VoIP.ms Call Forwarding:''' [[Call Forwarding]]

Hello Gubby

2026-08-28T18:50:25Z

Scomtois: Created page with "500px Hello Gubby is an AI receptionist platform that grows into the AI front desk for your business. '''Your AI receptionist answers ever..."

[[File:01-hello-gubby-logo.png|right|500px]]
Hello Gubby is an AI receptionist platform that grows into the AI front desk for your business.

'''Your AI receptionist answers every call in seconds with natural, human-sounding voices.''' It greets callers with your business name or a greeting of your choice, and answers questions about your services. It captures lead details and books appointments. When a caller needs a person, it warm transfers the call to you or your team, with unlimited warm transfers and unlimited notification recipients at no extra cost.

Every call is captured. After each call, you receive a summary, transcript, and recording on your email, in your Hello Gubby account, and in your CRM. Connect GoHighLevel or HubSpot directly from your Hello Gubby account, or sync your calendar for bookings. Webhooks are live, and Zapier is coming soon.

Hello Gubby is a no-code platform designed for busy business owners who need a done-for-you (DFY) setup at do-it-yourself (DIY) prices. Train your AI receptionist from your website or upload company materials, and go live in less than 5 minutes. When you are ready, add your AI receptionist to talk to your website visitors over voice or chat, and upload your own video or an AI receptionist video to give life to your website.

For more details or to sign up, visit: [https://hellogubby.ai hellogubby.ai]

__TOC__

== Before you start ==

To connect a number to Hello Gubby you will need:

# '''A VoIP.ms account with an active DID (phone number).''' If you don't have a number yet, follow: [[Order a DID Number]]
#:* You can leave the routing settings at their default values during the purchase. You will configure routing based on your use case (i.e., all calls answered by your AI receptionist, missed calls only, etc.) later in this guide.
# '''The DID on the pay as you go plan.'''
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Plan type</strong><br>Flat rate plans are limited to 2 simultaneous incoming calls, which limits how many callers your AI receptionist can answer at the same time. The pay as you go plan has no such restriction and is the recommended plan for AI receptionists. To change your plan, see the [[Changing DID Billing Plan]] article on this wiki.</div>
# '''A Hello Gubby account on a paid plan.''' Sign up at [https://hellogubby.ai hellogubby.ai]. Each plan comes with a 7-day free trial so try before you pay.

== Hello Gubby configuration ==

After you sign up, you can explore your Hello Gubby portal and test your AI receptionist before paying. You can set up your receptionist, modify its settings, and make a web call to listen to it, all without putting in your credit card. See this video walkthrough of the sign up and setup: [https://youtu.be/w0E3PMtBRIA?si=6CgC3OCARCjVKnys&t=360 How to sign up for Hello Gubby]

Once you pay, your subscription is active and you are navigated from Stripe back to your Hello Gubby portal, which takes you to your business number setup. You will see the You're live! confirmation, as shown below.

=== Connecting your number ===

# On the '''You're live!''' Confirmation screen, click on '''Continue'''.
#: [[File:hello-gubby-02-youre-live-continue.png|none|700px]]
# Choose how you want your AI front desk to answer calls: select '''Bring your own number''', then click on '''Continue'''.
#: [[File:hello-gubby-03-bring-your-own-number.png|none|700px]]
# You are taken to the '''SIP Trunk''' section of your settings. Click on the '''VoIP.ms''' card.
#: [[File:hello-gubby-04-sip-trunk-voipms-card.png|none|700px]]
#:If you skipped this step after payment, you can reach it anytime from '''Settings >> SIP Trunk''' in your Hello Gubby dashboard.
# '''Your VoIP.ms phone number:''' Enter your VoIP.ms DID with your country code (E.g. +1 778 819 7943).
# Click on '''Connect'''.
#: [[File:hello-gubby-05-connect-voipms-number.png|none|700px]]

=== Copy your SIP address ===

Once your number is connected, Hello Gubby shows it as your AI front desk line and displays your SIP address, along with the steps to complete in VoIP.ms.

# Click on '''Copy''' next to the SIP address.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Copy the address exactly</strong><br>Paste the address in VoIP.ms exactly as shown, with your number at the front. A shortened address without the number still connects calls, but your AI receptionist cannot tell which number was dialed and will answer with a generic greeting instead of your business greeting.</div>
#:[[File:hello-gubby-06-copy-sip-address.png|none|700px]]

The next section walks you through the two steps in your VoIP.ms portal.

== VoIP.ms configuration ==

You will now use the SIP address you copied from Hello Gubby in two places in your VoIP.ms portal: first to create a SIP URI, then to route your number to it.

=== Create the SIP URI ===

# In your VoIP.ms portal, go to '''DID Numbers >> SIP URI's'''.
#:[[File:hello-gubby-07-voipms-sip-uri-menu.png|none|700px]]
# Click on '''Add SIP uri'''.
#:[[File:hello-gubby-08-add-sip-uri.png|none|700px]]
# '''SIP URI:''' Paste the address you copied from your Hello Gubby dashboard.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 The sip: prefix is already there</strong><br>The field starts with sip: on its left. Paste only the address as copied (E.g. 17788197943@sip.hellogubby.ai). Do not type sip: again.</div>
# '''Description:''' Any name of your choice (E.g. Hello Gubby AI Receptionist).
# '''CallerID Override:''' Select '''I use a system capable of passing its own CallerID'''.
# '''CallerID E164:''' Leave as '''Disabled'''.
# Click on '''Save'''.
#:[[File:hello-gubby-09-create-sip-uri-form.png|none|700px]]

Your SIP URI is created and appears in the list.

: [[File:10-sip-uri-created.png|none|700px]]

=== Open your DID settings ===

# Go to '''DID Numbers >> Manage DID(s)'''.
#:[[File:hello-gubby-11-manage-dids-menu.png|none|700px]]
# Find your VoIP.ms number and click on the '''Edit DID''' (pencil) icon.
#:[[File:hello-gubby-12-edit-did.png|none|700px]]

=== Update the DID routing ===

# Under '''Routing Settings''', select '''SIP URI'''.
# In the dropdown next to it, choose the SIP URI you created.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Select the radio button, not just the dropdown</strong><br>Your SIP URI appearing in the dropdown is not enough. The round radio button next to SIP URI must be selected, or calls will keep going to the previous destination.</div>
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 This option routes ALL calls to your AI receptionist</strong><br>With SIP URI selected as your Main routing, every call to this number is answered by your AI receptionist. If you want your phone to ring first and your AI receptionist to answer only missed calls, or other setups, see '''Choosing when your AI receptionist answers''' below.</div>
#: [[File:hello-gubby-13-route-did-to-sip-uri.png|none|700px]]
# Scroll down and click on '''Apply Changes'''.

That's it! Every call to your VoIP.ms number is now answered by your AI receptionist. To have your AI receptionist answer only some calls instead, continue to the routing options below.

== Test your number ==

Call your VoIP.ms number from any phone.

Your AI receptionist should answer within a few seconds with your business greeting, for example: "Thanks for calling Bright Nest Cleaning, this is Filzah. How can I help you today?"

Then check your Hello Gubby dashboard: the call appears in your '''Call Log Inbox''' within a minute, with the summary, transcript, and recording.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 A generic greeting means the address was shortened</strong><br>If your AI receptionist answers with a generic Hello Gubby greeting instead of your business greeting, the SIP URI in VoIP.ms is missing your number at the front. Edit it under '''DID Numbers >> SIP URI's''' so it exactly matches the address shown in your Hello Gubby dashboard, with your number in front.</div>

== Choosing when your AI receptionist answers ==

The setup above routes every call to your AI receptionist. VoIP.ms routing gives you other options depending on how you want to run your business. All of them are configured on the same '''Edit DID Settings''' page, and none of them require any change in your Hello Gubby account.

=== Your AI receptionist answers every call ===

The default setup from this guide: '''Main routing = SIP URI'''. Your AI receptionist answers every call in seconds, 24/7, and warm transfers callers to you or your team when needed. Best for businesses that miss calls during jobs, after hours, or at busy times, and want every call captured.

=== Your phone rings first, your AI receptionist answers missed calls ===

Your phone rings normally, and your AI receptionist only picks up the calls you can't take. No call ever goes to voicemail again.

On your DID's '''Routing Settings''', the tabs across the top control where calls go in each situation:

'''4. Main:''' set to where your phone is. Select '''SIP/IAX''' if you use a softphone registered to your VoIP.ms account, or '''Call Forwarding''' with your mobile number (create the forwarding entry first under '''DID Numbers >> Call Forwarding''').

: [[File:hello-gubby-14-main-routing-phone.png|none|700px]]

'''5. If Busy:''' select '''SIP URI''' and choose the one you created.

'''6. If Unreachable:''' select '''SIP URI''' and choose the one you created.

'''7. If No Answer:''' select '''SIP URI''' and choose the one you created.

: [[File:hello-gubby-15-failover-routing-sip-uri.png|none|700px]]

'''8.''' Scroll down to '''Ring Time in seconds''' and set it to '''20 s (~4 Rings)'''. This is how long your phone rings before your AI receptionist takes the call.

'''9.''' Click on '''Apply Changes'''.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Watch out for your phone's own voicemail</strong><br>If your Main destination is call forwarding to a mobile phone, the mobile carrier's voicemail competes with this setup. If your carrier voicemail answers at 18 seconds and your VoIP.ms Ring Time is 20, voicemail takes every missed call before your AI receptionist can. Set your Ring Time shorter than your phone's voicemail delay, or disable your carrier voicemail. Softphone setups do not have this issue.</div>

Now your phone rings first on every call. If you don't answer within the ring time, are on another call, or your phone is offline, your AI receptionist answers and captures the lead. For more details on forwarding entries, see: [[Call Forwarding]]

=== Business hours to your phone, after hours to your AI receptionist ===

Your team answers during the day, and your AI receptionist covers evenings, weekends, and holidays.

# Go to '''DID Numbers >> Time Conditions''' and click on '''Create a new Time Condition'''.
# Set the time and day criteria for your business hours.
#:<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 VoIP.ms servers run on Eastern Time</strong><br>All VoIP.ms servers are set to Eastern Time, regardless of your location. Enter your hours converted to ET. For example, a Vancouver business open 9:00 to 17:00 Pacific enters 12:00 to 20:00. Compare your local time with the Current server time shown on the page.</div>
: [[File:hello-gubby-16-time-conditions-hours.png|none|700px]]
# '''Destination if one of the conditions matches server time (during business hours):''' select where your phone is, for example '''SIP/IAX''' or '''Call Forwarding'''.
# '''Destination if no conditions matches server time (after hours):''' select '''SIP URI''' and choose the one you created.
# '''Name of Time Condition:''' Any name of your choice (E.g. Business Hours). Click on '''Save Time Condition'''.
#: [[File:hello-gubby-17-time-conditions-destinations.png|none|700px]]
# Go to '''DID Numbers >> Manage DID(s)''', edit your number, and under '''Routing Settings''' select '''Time Conditions''' with the condition you created. Click on '''Apply Changes'''.
#: [[File:hello-gubby-18-select-time-condition-on-did.png|none|700px]]

Statutory holidays are not automatic: a time condition is a weekly schedule, so on holidays your phone still receives calls during the configured hours unless you temporarily adjust the condition. For more details, see: [[Time Conditions]]

=== Keep your existing main line and forward it to your AI receptionist ===

If your business number is with another provider and you don't want to move it, you can still use your AI receptionist. Set up call forwarding on your existing line (from your current provider's settings) to your VoIP.ms number, either for all calls or only when you don't answer. Your VoIP.ms number then routes to your AI receptionist as configured in this guide. Your published number never changes.

<div style="border:2px solid #d9534f; padding:8px 10px; margin:10px 0;"><strong>🚨 Check the caller number after your first forwarded call</strong><br>Some carriers replace the caller's number with the forwarding number when they forward a call. Make one test call and check the caller number shown in your Hello Gubby call log. Also note that forwarded calls are billed by both providers: your carrier's forwarding rate plus VoIP.ms per-minute usage.</div>

== Outbound calls and transfers ==

Warm transfers to your team and outbound calls from your VoIP.ms number: '''Coming soon!'''

== Useful resources ==

* '''Hello Gubby:''' [https://hellogubby.ai hellogubby.ai]
* '''Hello Gubby sign up and setup video:''' [https://youtu.be/w0E3PMtBRIA?t=360 YouTube walkthrough]
* '''VoIP.ms SIP URI feature:''' [[SIP URI]]
* '''VoIP.ms DID ordering:''' [[Order a DID Number]]
* '''VoIP.ms DID plans:''' [[Changing DID Billing Plan]]
* '''VoIP.ms Time Conditions:''' [[Time Conditions]]
* '''VoIP.ms Call Forwarding:''' [[Call Forwarding]]

File:Hello-gubby-18-select-time-condition-on-did.png

2026-08-28T18:33:09Z

Scomtois:

File:Hello-gubby-17-time-conditions-destinations.png

2026-08-28T18:33:03Z

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File:Hello-gubby-16-time-conditions-hours.png

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File:Hello-gubby-15-failover-routing-sip-uri.png

2026-08-28T18:32:53Z

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File:Hello-gubby-14-main-routing-phone.png

2026-08-28T18:32:41Z

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File:Hello-gubby-13-route-did-to-sip-uri.png

2026-08-28T18:32:28Z

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File:Hello-gubby-12-edit-did.png

2026-08-28T18:32:24Z

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File:Hello-gubby-11-manage-dids-menu.png

2026-08-28T18:32:19Z

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File:10-sip-uri-created.png

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File:Hello-gubby-09-create-sip-uri-form.png

2026-08-28T18:32:10Z

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File:Hello-gubby-08-add-sip-uri.png

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File:Hello-gubby-07-voipms-sip-uri-menu.png

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File:Hello-gubby-06-copy-sip-address.png

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File:Hello-gubby-05-connect-voipms-number.png

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File:Hello-gubby-04-sip-trunk-voipms-card.png

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File:Hello-gubby-03-bring-your-own-number.png

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File:Hello-gubby-02-youre-live-continue.png

2026-08-28T18:25:36Z

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File:02-youre-live-continue.png

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File:01-hello-gubby-logo.png

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File:Hello-Gubby-Logo.png

2026-08-27T21:02:22Z

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Textable

2026-08-04T15:38:34Z

Scomtois: /* Enable SMS and SMS URL callback for your DID */

[[File:Textable_logo2.png|250px|thumb|right|Textable]]

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Textable_FR Français] || [https://wiki.voip.ms/article/Textable_ES Español]
|}

Sign up to Textable: [https://textable.app/login?via=VoIPms Sign up link]

Textable is a messaging platform for VoIP phone numbers used by individuals, businesses and Managed Service Providers.<br><br>

<strong>Keep in touch with your customers – on the go</strong>

Your customers are constantly on the move and so are you. Give your customers the flexibility they need to keep in touch with you while you’re on the go.

<strong>Text Message from any browser</strong>

Real-time messaging application in your browser window with push notifications and always in sync with your mobile app.

<strong>Features that keep you connected with your customers as:</strong>

<ul>
<li>MMS Support</li>
<li>Organized Contacts</li>
<li>Canned Responses</li>
<li>Auto-Reply</li>
<li>Reminders</li>
<li>Drip Campaigns</li>
<li>Multi-Number Support</li>
<li>Away Messages</li>
</ul>

== Connecting Textable to your VoIP.ms DID ==

=== Enable the VoIP.ms API ===

<ol>
<li>First, you should login to your customer portal</li>
<li>Once logged in, head to Main menu>>SOAP and REST / JSON API at https://voip.ms/m/api.php</li>
<li>Here, you will need to set your API Password and Enable API. You will also need to disable the IP Whitelist by entering in 0.0.0.0 and clicking “Save IP Addresses”</li>
</ol>

[[File:Enabling_VoIP.ms_API.png|800px|thumb|left|Enabling the VoIP.ms API]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Enable SMS and SMS URL callback for your DID ===

We must now configure the inbound webhooks for incoming messages.

<ol>
<li>Go to DID numbers>>Manage DIDs in your customer portal (https://voip.ms/m/managedid.php)</li>
<li>Click the pencil icon to edit the DID number that you would like to use with Textable.</li>
<li>Under the Short Message Service section, check the “Enable SMS/MMS” and '''“SMS URL Callback”''' box.</li>
<li>Set the value of the input to '''https://inbound.textable.app/receive?provider=voipms&to={TO}&from={FROM}&msg={MESSAGE}&media={MEDIA}&id={ID}'''</li>
</ol>

'''<nowiki>https://inbound.textable.app/receive?provider=voipms&to={TO}&from={FROM}&msg={MESSAGE}&media={MEDIA}&id={ID}</nowiki>'''

'''Note: If you are a private label customer, please replace “https://inbound.textable.app” with your private label hostname
(e.g. '''“<nowiki>https://privatelabel-txb.textable.app/receive?provider=voipms......</nowiki>”''').
You should have received this value in your onboarding e-mail, but can also be requested by emailing support@textable.co.'''

You can optionally select the URL Retry option. This will do a continuous retry if the message failed to hit Textable' servers properly.
This may result in duplicate inbound messages from time to time.

[[File:Enabling_SMS_for_DID.png|800px|thumb|left|Enabling SMS & SMS URL Callback]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Textable configuration ===

In the Textable configuration interface, edit the user profile by choosing VoIP.ms as the provider and adding the API username and API password that you set up in the VoIP.ms customer portal.

Save the user details when done.

[[File:Textable_User_Details.png|800px|thumb|left|Textable configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Integrations ==

If you make use of services like Hubspot, MS Teams or Slack, Textable will allow you to integrate and use them with your VoIP.ms SMS service. To see how please see the instructions below.

=== HubSpot ===

'''Step 1:''' Create a Private App in HubSpot.
<br>To do this, in HubSpot click on the '''Settings''' button, and then go to <i>Integrations>>Private Apps</i>, then click on '''Create a private'''.
<br>On the '''Basic info tab''', enter a name and a description for your App.

[[File:Create_private_App_hubspot_Basic_Info.png|600px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Then, click on the tab '''Scopes''' and add the following items:

<ul>
<li>crm.objects.contact.read</li>
<li>crm.objects.contact.write</li>
<li>conversations.read</li>
<li>conversations.write</li>
</ul>

[[File:Create_private_App_hubspot_Scopes.png|600px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Once you have added the scopes, click on the button '''Create App'''.

'''Step 2:''' Get the App's access token.
<br>After creating the App, will be prompted a window where you can see the App access token. Click on "show token" and then on "copy".

[[File:Private_App_token.png|600px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Paste the token in a notepad since you will need it to connect Textable.

If you're not prompted a window with the token, you can go back to the Private Apps page and click on '''View access token'''.

'''Step 3:''' Integrate HubSpot into Textable by using the token of your Private Access App.
<br>Log into your Textable account. On the left menu, go to '''Integrations''', locate Hubspot and click on the connector button.

[[File:Textable_integrations_hubspot.png|700px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

On the window that will be prompted, set the following.

<ol>
<li>'''Select an installation option:''' Private App.</li>
<li>'''Name:''' Any name of your choice.</li>
<li>'''Webhook URL:''' Paste the URL of the Private App Access Token.</li>
</ol>

Click '''Ok''' when done.

[[File:Installing_Hubsport_Private_App_in_Textable.png|700px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

And that's it. Now HubSpot is ready to receive messages from the VoIP.ms DID that is configured on Textable.

=== MS Teams ===

'''Step 1:''' Add the App to the Team that will receive the messages from Textable.
<br>For this, in MS Teams go to Apps, search for '''Incoming Webhook''', click on it and then click on '''Add to a team'''. Select the team and then click on '''Setup connector'''.

[[File:Incoming_webhook_Add_to_Team.png|600px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

'''Step 2:''' Configure the App connector for the channel of the team.
<br>Go to the team and select the channel that will receive the messages (You can create a new one if needed), then click on '''More options''' and click on '''Connectors'''. Then, search for '''Incoming Webhook''' and click on '''Configure'''.

[[File:Connector_for_channel.png|200px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Set a name for the incoming webhook and then click on '''Create'''.

[[File:Incoming_webhook_App_Configure_Textable.png|600px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

After clicking on '''Create''' an URL will be generated. Copy and paste this into a notepad because you will need it.

'''Step 3:''' Integrate MS Teams into Textable by using the Webhook URL.
<br>Log into your Textable account. On the left menu, go to '''Integrations''', locate MS Teams and click on the connector icon.

[[File:Textable_integrations.png|700px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

On the window that will be prompted, set the following.

<ol>
<li>'''Select an installation option:''' Webhook.</li>
<li>'''Name:''' Any name of your choice.</li>
<li>'''Webhook URL:''' Paste the URL that was generated in Step 3.</li>
</ol>

Click '''Ok''' when done.

[[File:Install_MS_Teams.png|700px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

And that's it. Now your MS Teams is ready to receive messages from the VoIP.ms DID that is configured on Textable.

=== Slack ===

'''Step 1:''' Create a new App in Slack. Use this link if you don't know where is this option: [https://api.slack.com/apps?new_app=1 Create new App]
<br>Pick a name, choose a workspace to associate your app with, and then click '''Create App'''.

Bear in mind that you'll probably be receiving a lot of messages, so you might want to create a channel for Textable too.

[[File:Creating_new_Slack_App.png|500px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

'''Step 2:''' Enable Incoming Webhooks.
<br>After creating your new App, you'll be redirected to the settings page of this. From here select the Incoming Webhooks feature, and click the '''Activate Incoming Webhooks''' toggle to switch it on.

[[File:Activate_incoming_webhooks_for_Slack_App.png|500px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Now that Incoming Webhooks are enabled, the settings page should refresh and some extra options will appear. One of those options will be a really helpful button marked '''Add New Webhook to Workspace''', and you should click it.

[[File:Add_new_webhook_for_Slack_App.png|500px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Go ahead and pick a channel that the app will post to, and then click to '''Authorize''' your app.

[[File:Authorize_webhook_for_Slack.png|500px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

After authorizing the App you'll be sent back to your app settings, and you should now see a new entry under the Webhook URLs for your Workspace section, with a '''Webhook URL''' as can be seen in the picture below.

[[File:Webhook_URL.png|500px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Copy the URL because you will need it to configure Textable.

'''Step 3:''' Integrate Slack into Textable by using the Webhook URL.
<br>Log into your Textable account. On the left menu, go to '''Integrations''', locate Slack and click on the connector icon.

[[File:Textable_integrations_slacknew.png|700px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

On the window that will be prompted, set the following.

<ol>
<li>'''Select an installation option:''' Webhook.</li>
<li>'''Name:''' Any name of your choice.</li>
<li>'''Webhook URL:''' Paste the URL that was generated in Step 4.</li>
</ol>

Click '''Ok''' when done.

[[File:Installing_Slack_Webhook_in_Textable.png|700px|thumb|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

And that's it. Now your Slack channel is ready to receive messages from the VoIP.ms DID that is configured on Textable.

==Guide Links==

Textable guide : [https://www.textable.co/docs/voip-ms-setup-guide/ Setup Textable with VoIP.ms]

Textable support : [https://www.textable.co/contact-us/ Contact Textable support]

MS Teams integration: [https://learn.microsoft.com/en-us/microsoftteams/platform/webhooks-and-connectors/how-to/add-incoming-webhook Create incoming Webhooks]

Slack integration: [https://api.slack.com/messaging/webhooks Sending messages using Incoming Webhooks]

HubSpot integration: [https://developers.hubspot.com/docs/api/private-apps#create-a-private-app Create a private App]

Mitel 6900 Series SIP Phones

2026-07-07T19:53:30Z

Scomtois:

[[File:Mitel_6900_Series.png|right|none|Mitel 6900 Series IP Phones]]
This guide explains how to configure a Mitel 6900 Series Open SIP phones with VoIP.ms.
Please note that menu names and available options may vary depending on the phone model, firmware version, and provisioning method being used.

= Supported Models =
This guide applies generally to Mitel 6800 and 6900 Series IP phones, including but not limited to:
:* 6863i
:* 6865i
:* 6867i
:* 6869i
:* 6873i
:* 6905
:* 6907
:* 6910
:* 6915
:* 6920/6920w
:* 6930/6930w
:* 6940/6940w
:* 6970
:* M695 PKM ''(If supported)''

= Firmware Note =
<div style="position: relative; width: 90%; /* height: 20px; */ background-color: #f0fbff; border: 2px solid #2f6185; border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; flex-wrap: wrap; text-align: stretch; font-family: sans-serif; color: #333;">
<div style="position: absolute; margin-top:35px; margin-left: 20px; font-size: 150px; opacity: 0.1; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-20deg);">ℹ️</div>
<p style="position: relative; z-index: 2; margin: 0; line-height: 1.5;">

:'''The phone must be running SIP firmware.'''
:'''If the phone was previously used with another platform or PBX, make sure it is running SIP firmware before attempting to register it with VoIP.ms.'''
:'''If the phone is not running SIP firmware, please refer to the manufacturer documentation for firmware conversion or firmware upgrade instructions.'''
</p></div>

= Step 1 - Create or Verify the VoIP.ms Sub Account =
Access your VoIP.ms Customer Portal:
:# Go to <code>Sub Accounts</code>
:# Create a new '''Sub Account''' or edit an existing one
:# Take note of the '''SIP Username and SIP Password'''
:# Make sure the selected codec matches the codec configured on the phone
:# Save the changes
:::Recommended Sub Account settings:
::::'''Device Type:''' ATA Device, IP Phone or Softphone
::::'''SIP Username:''' Your Sub Account username <code>100000_office</code>
::::'''Password:''' Your SIP password
::::'''Protocol:''' SIP
::::'''Allowed Codecs:''' Select the codecs you will use on the Mitel phone

= Step 2 - Obtain the Phone IP Address =
Connect the phone to the network and power it on.
From the phone keypad or screen:
:# Press the <code>Settings</code> button
:# Navigate to <code>Status</code>
:# Go to <code>Network</code>
:# Locate the phone IP address

:The path may appear as:
:: <pre>Settings -> Status -> Network -> IP Address</pre>
:: On some models, the IP address may be located under:
:: <pre>Phone Status -> IP & MAC Addresses</pre>

= Step 3 - Access the Phone Web Interface =
From a computer connected to the same network as the phone:
:# Open a web browser
:# Enter the phone IP address in the address bar
:# Log in as administrator

Default administrator credentials are commonly:
: '''Username:''' <code>admin</code>
: '''Password:''' <code>22222</code>

'''Note:'''
:If these credentials do not work, the password may have been changed by a previous administrator. In that case, please refer to the device documentation or reset procedure.

'''Security recommendation:'''
:It is strongly recommended to change the default administrator password after completing the configuration.

= Step 4 - Configure SIP Line 1 =
Once logged into the phone Web Interface:
:# Go to <code>Advanced Settings</code>
:# Select <code>Line 1</code>

The SIP settings are usually divided into two main sections:
:* Basic SIP Authentication Settings
:* Basic SIP Network Settings

== Basic SIP Authentication Settings ==
Under <code>Basic SIP Authentication Settings</code>, configure the following fields:
:'''Screen Name:''' Name that displays on the idle screen
:'''Screen Name 2:''' Custom text message that displays on the idle screen.
:: [[File:Mitel_6900_ScreenName.png|none|350px]]
:'''Phone Number:''' Enter your VoIP.ms SubAccount Username <code>######_SubAccount</code>
:'''Authentication Name:''' Enter your VoIP.ms SubAccount Username <code>######_SubAccount</code>
:'''Password:''' Use the SIP password from the Main Account or Sub Account.
:'''Display Name:''' Enter your outbound Caller ID Name that will be displayed for Canadian Numbers

<div style="position: relative; width: 90%; /* height: 20px; */ background-color: #fff3cd; border: 2px solid #ffeeba; border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; flex-wrap: wrap; text-align: stretch; font-family: sans-serif; color: #856404;">
<div style="position: absolute; margin-top:35px; margin-left: 20px; font-size: 150px; opacity: 0.1; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-20deg);">⚠️</div>
<p style="position: relative; z-index: 2; margin: 0; line-height: 1.5;">

:- We suggest entering your outbound Caller ID Name must be in '''CAPITAL LETTERS'''. This will appears more clearly/visible on some devices.
:- You must '''NOT''' use any special characters, they will not be displayed.
:- '''Do not exceed 15 characters'''. Some of regular Canadian providers will '''not''' show more than 15 characters. We suggest shrinking or adapt your caller ID.
:- Spaces are allowed and count as a character.
</p></div>

:: [[File:Mitel_6900_Line1_Basic_SIP_Auth.png|none|500px]]

== Basic SIP Network Settings ==
Under <code>'''Basic SIP Network Settings'''</code>, configure the following fields:
:'''Proxy Server:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>''
:'''Proxy Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Outbound Proxy:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>''
:'''Proxy Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Registrar Server:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>''
:'''Registrar Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Registration Period:''' <code>120</code> ''secondes''
:: Recommended registration expiry value.

:: [[File:Mitel_6900_Line1_Basic_SIP_Network.png|none|500px]]

== Step 5 - Save the Configuration ==
After entering the '''SIP authentication''' and '''SIP network settings''':
:# Click''' <code>Save Settings</code>'''
:# Reboot the phone if required
:# Wait for the phone to register

The phone should show the line as registered once the configuration is successful.

= Codec Settings =
You can change the default codecs selection on the phone to match the codecs enabled in your VoIP.ms account or Sub Account.
: '''Note''': This is optional, as the Basic Codecs Includes G.711u.

= Firewall and NAT Considerations =
Recommended router settings:
:* Disable SIP ALG
:* Make sure the phone can resolve DNS hostnames
:* Make sure the phone can reach the selected VoIP.ms server
:* Avoid multiple SIP devices using the same local SIP port when possible

= Test Calls =
After the phone registers, test the configuration.
: VoIP.ms test numbers:
:* '''4443:''' Echo test
:* '''4747:''' DTMF test
:* '''822:''' Caller ID playback test
:* '''*98''' or '''*97''': Voicemail access

It is also recommended to test:
:* Outbound calls
:* Inbound calls
:* Two-way audio
:* DTMF through an IVR
:* Voicemail access

= Troubleshooting =
== The Phone Does Not Register ==
If the phone does not register, verify the following:
:* The SIP Username is correct
:* The SIP Password is correct
:* The correct VoIP.ms PoP server is being used
:* The SIP port is correct
:* The selected transport protocol matches the account configuration
:* SIP ALG is disabled on the router
:* The account or Sub Account is not blocked due to repeated failed registration attempts
:* If using SIP-TLS, encrypted SIP traffic is enabled in the VoIP.ms portal and the phone is configured for TLS/SRTP

== Incoming Calls Do Not Reach the Phone ==
If outbound calls work but inbound calls do not, verify the following:
:* The DID is routed to the correct SIP account or Sub Account
:* The DID PoP matches the server used by the phone
:* The phone is currently registered
:* Failover routing is not sending the call elsewhere
:* The firewall is not blocking SIP or RTP traffic

=== One-Way Audio ===
If one-way audio occurs, verify the following:
:* SIP ALG is disabled
:* RTP ports are not blocked by the firewall
:* The phone is not behind multiple NAT routers
:* The selected codec is enabled in both the phone and the VoIP.ms portal
:* The device is registering to the correct VoIP.ms server

=== DTMF Is Not Working ===
If DTMF tones are not detected properly:
:* Set DTMF to <code>RFC2833</code> or <code>AVT</code>
:* Avoid using in-band DTMF
:* Test using the VoIP.ms DTMF test number: <code>4747</code>

Mitel 6900 Series SIP Phones

2026-07-07T13:34:56Z

Scomtois: /* Supported Models */

[[File:Mitel_6900_Series.png|right|none|Mitel 6900 Series IP Phones]]
This guide explains how to configure a Mitel 6900 Series Open SIP phones with VoIP.ms.
Please note that menu names and available options may vary depending on the phone model, firmware version, and provisioning method being used.

= Supported Models =
This guide applies generally to Mitel 6800 and 6900 Series IP phones, including but not limited to:
:* 6863i
:* 6865i
:* 6867i
:* 6869i
:* 6873i
:* 6905
:* 6907
:* 6910
:* 6915
:* 6920/6920w
:* 6930
:* 6940
:* 6970
:* M695 PKM ''(If supported)''

= Firmware Note =
<div style="position: relative; width: 90%; /* height: 20px; */ background-color: #f0fbff; border: 2px solid #2f6185; border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; flex-wrap: wrap; text-align: stretch; font-family: sans-serif; color: #333;">
<div style="position: absolute; margin-top:35px; margin-left: 20px; font-size: 150px; opacity: 0.1; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-20deg);">ℹ️</div>
<p style="position: relative; z-index: 2; margin: 0; line-height: 1.5;">

:'''The phone must be running SIP firmware.'''
:'''If the phone was previously used with another platform or PBX, make sure it is running SIP firmware before attempting to register it with VoIP.ms.'''
:'''If the phone is not running SIP firmware, please refer to the manufacturer documentation for firmware conversion or firmware upgrade instructions.'''
</p></div>

= Step 1 - Create or Verify the VoIP.ms Sub Account =
Access your VoIP.ms Customer Portal:
:# Go to <code>Sub Accounts</code>
:# Create a new '''Sub Account''' or edit an existing one
:# Take note of the '''SIP Username and SIP Password'''
:# Make sure the selected codec matches the codec configured on the phone
:# Save the changes
:::Recommended Sub Account settings:
::::'''Device Type:''' ATA Device, IP Phone or Softphone
::::'''SIP Username:''' Your Sub Account username <code>100000_office</code>
::::'''Password:''' Your SIP password
::::'''Protocol:''' SIP
::::'''Allowed Codecs:''' Select the codecs you will use on the Mitel phone

= Step 2 - Obtain the Phone IP Address =
Connect the phone to the network and power it on.
From the phone keypad or screen:
:# Press the <code>Settings</code> button
:# Navigate to <code>Status</code>
:# Go to <code>Network</code>
:# Locate the phone IP address

:The path may appear as:
:: <pre>Settings -> Status -> Network -> IP Address</pre>
:: On some models, the IP address may be located under:
:: <pre>Phone Status -> IP & MAC Addresses</pre>

= Step 3 - Access the Phone Web Interface =
From a computer connected to the same network as the phone:
:# Open a web browser
:# Enter the phone IP address in the address bar
:# Log in as administrator

Default administrator credentials are commonly:
: '''Username:''' <code>admin</code>
: '''Password:''' <code>22222</code>

'''Note:'''
:If these credentials do not work, the password may have been changed by a previous administrator. In that case, please refer to the device documentation or reset procedure.

'''Security recommendation:'''
:It is strongly recommended to change the default administrator password after completing the configuration.

= Step 4 - Configure SIP Line 1 =
Once logged into the phone Web Interface:
:# Go to <code>Advanced Settings</code>
:# Select <code>Line 1</code>

The SIP settings are usually divided into two main sections:
:* Basic SIP Authentication Settings
:* Basic SIP Network Settings

== Basic SIP Authentication Settings ==
Under <code>Basic SIP Authentication Settings</code>, configure the following fields:
:'''Screen Name:''' Name that displays on the idle screen
:'''Screen Name 2:''' Custom text message that displays on the idle screen.
:: [[File:Mitel_6900_ScreenName.png|none|350px]]
:'''Phone Number:''' Enter your VoIP.ms SubAccount Username <code>######_SubAccount</code>
:'''Authentication Name:''' Enter your VoIP.ms SubAccount Username <code>######_SubAccount</code>
:'''Password:''' Use the SIP password from the Main Account or Sub Account.
:'''Display Name:''' Enter your outbound Caller ID Name that will be displayed for Canadian Numbers

<div style="position: relative; width: 90%; /* height: 20px; */ background-color: #fff3cd; border: 2px solid #ffeeba; border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; flex-wrap: wrap; text-align: stretch; font-family: sans-serif; color: #856404;">
<div style="position: absolute; margin-top:35px; margin-left: 20px; font-size: 150px; opacity: 0.1; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-20deg);">⚠️</div>
<p style="position: relative; z-index: 2; margin: 0; line-height: 1.5;">

:- We suggest entering your outbound Caller ID Name must be in '''CAPITAL LETTERS'''. This will appears more clearly/visible on some devices.
:- You must '''NOT''' use any special characters, they will not be displayed.
:- '''Do not exceed 15 characters'''. Some of regular Canadian providers will '''not''' show more than 15 characters. We suggest shrinking or adapt your caller ID.
:- Spaces are allowed and count as a character.
</p></div>

:: [[File:Mitel_6900_Line1_Basic_SIP_Auth.png|none|500px]]

== Basic SIP Network Settings ==
Under <code>'''Basic SIP Network Settings'''</code>, configure the following fields:
:'''Proxy Server:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>''
:'''Proxy Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Outbound Proxy:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>''
:'''Proxy Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Registrar Server:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>''
:'''Registrar Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Registration Period:''' <code>120</code> ''secondes''
:: Recommended registration expiry value.

:: [[File:Mitel_6900_Line1_Basic_SIP_Network.png|none|500px]]

== Step 5 - Save the Configuration ==
After entering the '''SIP authentication''' and '''SIP network settings''':
:# Click''' <code>Save Settings</code>'''
:# Reboot the phone if required
:# Wait for the phone to register

The phone should show the line as registered once the configuration is successful.

= Codec Settings =
You can change the default codecs selection on the phone to match the codecs enabled in your VoIP.ms account or Sub Account.
: '''Note''': This is optional, as the Basic Codecs Includes G.711u.

= Firewall and NAT Considerations =
Recommended router settings:
:* Disable SIP ALG
:* Make sure the phone can resolve DNS hostnames
:* Make sure the phone can reach the selected VoIP.ms server
:* Avoid multiple SIP devices using the same local SIP port when possible

= Test Calls =
After the phone registers, test the configuration.
: VoIP.ms test numbers:
:* '''4443:''' Echo test
:* '''4747:''' DTMF test
:* '''822:''' Caller ID playback test
:* '''*98''' or '''*97''': Voicemail access

It is also recommended to test:
:* Outbound calls
:* Inbound calls
:* Two-way audio
:* DTMF through an IVR
:* Voicemail access

= Troubleshooting =
== The Phone Does Not Register ==
If the phone does not register, verify the following:
:* The SIP Username is correct
:* The SIP Password is correct
:* The correct VoIP.ms PoP server is being used
:* The SIP port is correct
:* The selected transport protocol matches the account configuration
:* SIP ALG is disabled on the router
:* The account or Sub Account is not blocked due to repeated failed registration attempts
:* If using SIP-TLS, encrypted SIP traffic is enabled in the VoIP.ms portal and the phone is configured for TLS/SRTP

== Incoming Calls Do Not Reach the Phone ==
If outbound calls work but inbound calls do not, verify the following:
:* The DID is routed to the correct SIP account or Sub Account
:* The DID PoP matches the server used by the phone
:* The phone is currently registered
:* Failover routing is not sending the call elsewhere
:* The firewall is not blocking SIP or RTP traffic

=== One-Way Audio ===
If one-way audio occurs, verify the following:
:* SIP ALG is disabled
:* RTP ports are not blocked by the firewall
:* The phone is not behind multiple NAT routers
:* The selected codec is enabled in both the phone and the VoIP.ms portal
:* The device is registering to the correct VoIP.ms server

=== DTMF Is Not Working ===
If DTMF tones are not detected properly:
:* Set DTMF to <code>RFC2833</code> or <code>AVT</code>
:* Avoid using in-band DTMF
:* Test using the VoIP.ms DTMF test number: <code>4747</code>

3CX Phone System - v20

2026-07-06T19:48:34Z

Scomtois: /* Sub-Account Setup */

[[File:3CX_Logo.jpg|300px|right|link=https://www.3cx.com/|3CX home page]]

3CX offers a complete Unified Communications solution out of the box.
As an open standards software PBX, installation and management of your phone system have never been easier.

== Pre-requisites ==

=== Latest 3CX V20 Update ===
:- Ensure you have the '''latest update''' of 3CX V20 installed.

=== Sub-Account Setup ===
Before proceeding, create a sub-account with the following settings:
:- '''Authentication type''': (x) User/Password Authentication
:- '''Device type''': (x) Asterisk, IP PBX, Gateway or VoIP Switch
:- '''CallerID Number''': (x) I use a system capable of passing its own CallerID
:- '''Allowed Codecs''': Only '''select G711''' or you can select G722 for HD voice
:- '''rInstance Parameter''': Set to '''"Yes"'''

= Trunk Configuration =

If you do not already have set a VoIP.ms trunk in your instance, you will need to create one.
: [1] Log into your 3CX web interface.
: [2] Click the '''"Admin"''' icon located at the bottom left corner.
: [3] In the left menu, select '''"Voice & Chat"'''.
: [4] Click the '''"[+ Add Trunk]"''' button.

== General Tab ==

Before completing the fields, load the VoIP.ms Trunk template:
: [1] Under '''"General"''' tab, find the '''"SIP Trunk Details"''' section.
: [2] Country: Select '''"CA"'''.
: [3] Trunk: Select '''"VoIP.ms"'''.

Now configure the trunk:
: [1] '''Name''': Customize as needed.
: [2] '''Default Route''': Leave it as '''"End calls"'''.
: [3] '''Limit To''': Choose the user group to limit trunk access, default is '''"DEFAULT"'''.
: [4] '''Outbound Rules''': Check '''"[x] Create an outbound rules for this SIP Trunk"''' if no outbound route is created yet.
: [5] '''Registrar/Server''': Enter the POP server matching your DID settings.
: [6] '''Main Trunk Number''': Enter the default DID for this trunk.
: [7] '''Simultaneous Calls''': Set trunk limit based on your 3CX license.
: [8] '''Authentication ID (SIP User ID)''': Enter sub-account username: ''eg: #####_SubAccount''.
: [9] '''Authentication Password''': Enter sub-account password.

== DID Numbers Tab ==

If using multiple DIDs:
: [1] Go to the '''"DID Numbers"''' tab.
: [2] Click '''"[+ Add]"''' button.
: [3] Type VoIP.ms DID '''without hyphen, dot, or parentheses'''.
: [4] Press '''Enter'''.

== Options Tab ==

If setting up outbound caller ID Name:
: [1] Go to the '''"Options"''' tab.
: [2] Click '''"Caller ID Control"'''.
: [3] Enter your Caller ID Name by following the requirements '''below'''.
<div style="margin-left:50px;
padding:8px;
border-left: 6px solid rgb(73, 111, 199, 0.8);
box-shadow: 5px 4px 4px 4px #f5f5f5;
background-color: rgba(73, 111, 199, 0.08);
width:70%;">

'''IMPORTANT - Caller ID Name'''<br/>
* Entering outbound Caller ID Name, it must be in '''CAPITAL LETTERS'''. ''This will appears more clearly/visible on some devices.'' <br/>
* You must '''NOT use any special characters'''. ''They will not be displayed and it prevents potential outbound call issues.'' <br/>
* Do '''NOT exceed 15 characters'''. ''Traditional canadian providers will '''not show more than 15 characters'''. We suggest shrinking or adapt the name.''<br/>
* Spaces are allowed in a caller ID name but count as a characters. ''(do not exceed 15 chars. in total)''.

</div>

Click '''"[SAVE]"''' button at the top.

== SMS Tab ==
===Service Limitations===

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 56, 45, 0.08);
width:70vw;">

🏗️ ''' Temporary Limitation'''
: ➜ '''Temporarily''' the current integration supports a maximum of '''160 characters per sent message'''. We are currently working on a way to permit more than 160 characters and will be available soon.
: ➜ Messages longer than 160 characters will result in an error, noticeable by a ❌ where your message. The error can be retrieved in your 3CX Admin console under '''Event Log'''.
</div>

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITATIONS'''
: ➜ VoIP.ms supports ''Application-to-Person (A2P)'' delivery for business text messaging. However, requirements and fees vary due to industry regulations. The default limit is 100 SMS per day, which can be raised upon request. Contact us at '''messaging@voip.ms''' to start your verification process.
<br/>
: ➜ The limit of sending SMS/MMS using the '''Bearer token''' is limited to the '''same limit''' as our '''SOAP and REST/JSON API''' which is '''<u>100 per day</u>'''. For any modification to the limit, please contact our support team through the live chat or open a ticket via the customer portal.
: ➜ The SMS/MMS service is only available for <u>'''US and Canadian DID and Toll-Free Numbers'''</u> marked with the distinctive '''SMS/MMS Icon'''.
: ➜ The SMS/MMS service can only be used to send SMS/MMS Messages to <u>'''Canadian and US'''</u> 10 digit numbers at this time.
: ➜ We cannot guarantee that '''accents or special characters''' including non-Latin letters will be properly delivered.
: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''are widely restricted'''</font> by mobile providers as a response from the industry to combat SPAM campaigns and protect the consumer. Including this type of links into your messages '''<u>may result in a blocked or undelivered message</u>'''.
: ➜ VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS/MMS service from any account that may present marketing patterns/automation patterns/bulk sending or receiving.

</div>

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 NOTE '''
: When there is an error while sending a message, you will notice a ❌ where your message. You can retrieve the error details in your 3CX Admin console under '''Event Log'''.
</div>

===Generate a Bearer Token===

If you do not already have a "Bearer Token" generated that you have copied somewhere, click on "Generate New Token" to obtain one.
You will need to generate a new one.

: 1) Go to your VoIP.ms client portal '''"Main Menu"''' > '''"SOAP and REST/JSON API"'''
: 2) Click on '''"Enable"''' the API if its not already made.
: 3) Where '''Bearer Token''' click on '''Generate New Token'''

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 149, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

🔐 '''ATTENTION: Store your generated token securely!''' ⚠️
: When you generate a Bearer Token, be sure to copy and store it in a secure place, as it won't be visible again after generation.
: It will only be available to be copied once it being generated. If you lost your Bearer Token, you need to regenerate a new one.'''

</div>

=== Add your Bearer Token to your 3CX trunk ===
Once the trunk is created and configured
:1) Go to the '''"SMS"''' tab.
:2) '''Enable SMS''': [x] Enable the service by checking the checkbox.
:3) In the '''"API Key"''' field, paste your VoIP.ms ''[[#Generate a Bearer Token|generated Bearer Token]]''.

:: [[File:VOIPMS_3CX_V20_SMS_TAB.png|none]]

:4) Then, copy the Webhook URL, you can click on the '''copy 📋 icon '''.

=== Set the 3CX trunk Webhook URL to your VoIP.ms DID ===
:1) Go to your VoIP.ms client portal and navigate to '''"DID Number"''' tab > '''"Manage DID"''' >
:2) click the '''yellow Edit button''' of your DID you wish to use for SMS.
:3) Scroll down to the SMS section until you see the '''Message Service (SMS/MMS)''' section.
:4) '''Message Service (SMS/MMS)''': [x] '''Enable'''
:5) In the '''"SMS/MMS Webhook URL (3CX)"''' field, paste the '''Webhook URL''' from the new 3CX trunk.
:6) '''Dialing Mode for SMS (3CX)''': Select '''NANPA'''

:: [[File:3CXv20_DID_Edit_SMS.png|700px]]

Done! Simply associate the DID to your Queue or your extension.

=[[#SMS_Tab | Configuring VoIP.ms SMS/MMS service]]=
[[#SMS_Tab |Click here to see the section SMS/MMS Tab]]

Catalaya - Orchid Link SBC

2026-06-23T18:32:48Z

Scomtois: moved Catalaya - Orchid Link SBC to Cataleya - Orchid Link SBC

#REDIRECT [[Cataleya - Orchid Link SBC]]

Cataleya - Orchid Link SBC

2026-06-23T18:32:48Z

Scomtois: moved Catalaya - Orchid Link SBC to Cataleya - Orchid Link SBC

[[File:Cataleya logo.png|400px|right|none]]

This article provides general configuration guidelines for connecting an '''Orchid Link Session Border Controller (SBC)''' with [[VoIP.ms]] SIP trunking. The configuration described below focuses on two Orchid Link zones:

:* '''Access Public''': VoIP.ms-facing side of the SBC.
:* '''Access Core''': customer network, SIP platform, IP-PBX, softswitch, or core-facing side of the SBC.

'''Audience:''' Network and voice engineers familiar with SIP, VoIP.ms accounts, and basic Orchid Link administration.

'''Vendor confirmation pending:''' Supported Orchid Link software versions and deployment models should be confirmed with Cataleya before adding a component information table to this article.

'''Note:''' The screenshots and field values shown in this article are examples only. Adjust the configuration according to your own network design, security policy, licensing, and Orchid Link deployment.

'''The Microsoft Teams section is a high-level overview and may require additional configuration and testing.'''

__TOC__

= Prerequisites =

Before starting the configuration, make sure you have the following information available:

:* Access to the Orchid Link administrative interface.
:* A configured customer and partition in Orchid Link.
:* SIP and media interfaces already created for both the public and core sides.
:* The VoIP.ms server or POP information that will be used by the trunk.
:* The trusted VoIP.ms source IP addresses or POPs that will be allowed through the SBC security rules.
:* SIP credentials or trunk information required for the VoIP.ms side.
:* The internal IP address and SIP port of the customer-side PBX, softswitch, or SIP platform.
:* Any required SIP Parameters, Transparency, Media, Call Admission Control, and Routing Policy profiles.

= Configuration Overview =

The Orchid Link SBC configuration is separated into two main zones:

:* Configure the '''Access Public''' zone first. This zone represents the VoIP.ms-facing side of the SBC.
:* Configure the '''Access Core''' zone after the Access Public zone. This zone represents the customer network or PBX-facing side of the SBC.

Once both zones are created, routing is configured so that calls can pass between the VoIP.ms side and the core-side SIP environment.

= Access Public Zone Configuration =

The Access Public zone is the side of the SBC that communicates with VoIP.ms. This zone should be configured first.

== Step 1 - General ==

In the '''General''' tab, configure the mandatory fields shown in the Orchid Link interface.

[[File:Orchid_Link_Access_Public_General.png|thumb|600px|Access Public zone - General tab]]

:* '''Name''': Enter a descriptive name for the zone, for example, '''VoIPMS_Public'''.
:* '''Customer''': Select the customer associated with this zone.
:* '''Partition''': Select the partition allocated to this customer.
:* '''Zone Type''': Select '''AccessPublic'''.
:* '''Time Zone''': Select the appropriate time zone for the customer.
:* '''Trunk Group Type''': Leave this set to '''None''', unless your deployment requires another value.
:* '''Service Level''': Keep the default service level, for example, '''Gold''', unless your policy requires otherwise.

== Step 2 - Signaling ==

In the '''Signaling''' step, select the SIP-related profiles and interfaces used for the VoIP.ms-facing side.

[[File:Orchid_Link_Access_Public_Signaling.png|thumb|600px|Access Public zone - Signaling step]]

:* '''SIP Profile''': Select the preconfigured SIP Parameters Profile from the drop-down list.
::* This profile is created under '''Profiles > SIP Parameters'''.

[[File:Orchid_Link_SIP_Parameters_Profile.png|thumb|600px|SIP Parameters Profile example]]

:* '''Transparency Profile''': Select the transparency profile created for this deployment.
::* This profile can be created under '''Profiles > Transparency'''.

[[File:Orchid_Link_Transparency_Profile.png|thumb|500px|Transparency Profile example]]

:* '''SIP Interface''': Select the SIP Interface configured for the public access side and add it to the zone.

== Step 3 - Media ==

In the '''Media''' step, select the media interface and media profile used for VoIP.ms traffic.

[[File:Orchid_Link_Access_Public_Media.png|thumb|600px|Access Public zone - Media step]]

:* '''Media Interface''': Select the Media Interface configured for the public access side and add it to the zone.
:* '''Media Profile''': Select the Media Profile used for VoIP.ms traffic.
::* The Media Profile is created under '''Profiles > Media'''.

[[File:Orchid_Link_Media_Profile.png|thumb|500px|Media Profile example]]

== Step 4 - Call Admission Control ==

In the '''Call Admission Control''' step, select the CAC profile that will control the number of concurrent sessions allowed for this zone.

[[File:Orchid_Link_CAC_Profile_Zone.png|thumb|450px|Zone CAC Profile example]]

:* '''CAC Profile Zone''': Select the CAC profile created for this zone.
::* The profile is created under '''Profiles > Call Admission Control'''.
::* Configure the session count according to the licensed capacity allocated for VoIP.ms traffic.
:* '''CAC Profile UE''': Select the UE CAC profile if you want to apply per-endpoint limits.

[[File:Orchid_Link_CAC_Profile_UE.png|thumb|450px|UE CAC Profile example]]

== Step 5 - Security ==

In the '''Security''' step, configure the allowed sources and any required security parameters according to your security policy.

[[File:Orchid_Link_Access_Public_Security.png|thumb|600px|Access Public zone - Security step]]

:* Restrict access to known and trusted VoIP.ms IP addresses or POPs whenever possible.
:* Avoid allowing unrestricted public SIP access unless this is specifically required and properly secured.
:* Apply any additional SBC security settings required by your deployment.

== Step 6 - Remote Peers ==

No configuration is required in this step for the Access Public zone.

:* Leave the default values unless a specific deployment requires otherwise.

== Step 7 - Routing ==

In the '''Routing''' step, define where traffic from the Access Public zone should be sent.

[[File:Orchid_Link_Access_Public_Routing.png|thumb|600px|Access Public zone - Routing step]]

:* '''Egress Zone''': Select the '''Access Core''' zone as the egress destination for this Access Public zone.

== Step 8 - Transcoding ==

If transcoding is not required, no configuration is needed in this step.

:* Configure transcoding only if codec interworking is required between VoIP.ms and the customer-side SIP environment.
:* The transcoding profile should match your network design and licensing.

= Access Core Zone Configuration =

The Access Core zone represents the customer network or PBX-facing side of the SBC. Configure this zone after the Access Public zone has been created.

== Step 1 - General ==

In the '''General''' tab, configure the core-side zone information.

:* '''Zone Type''': Select '''AccessCore'''.
:* '''Name''': Use a clear name that identifies the customer or core-side environment.
:* '''Customer''': Select the relevant customer.
:* '''Partition''': Select the partition allocated to the customer.
:* '''Time Zone''': Select the appropriate time zone.
:* '''Trunk Group Type''': Leave as '''None''', unless your deployment requires otherwise.
:* '''Service Level''': Use the appropriate service level for the customer or deployment.

== Step 2 - Signaling ==

In the '''Signaling''' step, configure the SIP interface and profiles used toward the customer core network.

:* '''SIP Interface''': Select the SIP Interface configured for the core side.
:* '''SIP Profile''': Select the appropriate SIP Parameters Profile for the core network.
::* This profile may be different from the profile used on the VoIP.ms-facing side.
:* '''Transparency Profile''': Select the appropriate transparency profile.
::* This profile can be created under '''Profiles > Transparency'''.

== Step 3 - Media ==

In the '''Media''' step, configure the media interface and media profile used toward the core network.

:* '''Media Interface''': Select the Media Interface configured for the core side.
:* '''Media Profile''': Select the Media Profile used for traffic toward the core network (customer network or PBX).

== Step 4 - Call Admission Control ==

In the '''Call Admission Control''' step, select the CAC profile used for the core-facing side.

:* '''CAC Profile Zone''': Select the core-side CAC Profile Zone.
::* Configure the session count according to the capacity you want to reserve for core-side calls.
:* '''CAC Profile UE''': Select the UE CAC profile if per-endpoint limits are required on the core side.

== Step 5 - Security ==

In the '''Security''' step, configure the allowed source addresses for the core-side signaling devices.

:* Only allow traffic from trusted internal networks, or core devices
:* Do not expose the core-side zone to untrusted networks unless this is part of the intended design and is properly secured.

== Step 6 - Remote Peers ==

In the '''Remote Peers''' step, configure the customer-side SIP endpoint information.

[[File:Orchid_Link_Access_Core_Remote_Peers.png|thumb|600px|Access Core zone - Remote Peers step]]

:* '''Remote Endpoints''': Configure the IP address and port of the remote core endpoint.
::* Examples include an IP-PBX, softswitch, SIP application server, or another SBC.
::* The Orchid Link SBC will send SIP signaling to these addresses.

== Step 7 - Routing ==

In the '''Routing''' step, select the routing policy that will be used for traffic toward the core side.

[[File:Orchid_Link_Access_Core_Routing.png|thumb|600px|Access Core zone - Routing step]]

:* '''Routing Policy''': Select the appropriate Session Routing Policy from the drop-down list.
::* The policy is configured under '''Policy Configuration > Session Routing Policies'''.

[[File:Orchid_Link_Session_Routing_Policy.png|thumb|600px|Session Routing Policy example]]

== Step 8 - Transcoding ==

Configure transcoding only if codec interworking is required between VoIP.ms and your core network.

:* If transcoding is not required, no changes are needed in this step.
:* If transcoding is required, select or create the appropriate transcoding profile according to the deployment requirements.

= Final Checklist =

After completing the configuration, review the following items:

:* The Access Public zone is configured as '''AccessPublic'''.
:* The Access Core zone is configured as '''AccessCore'''.
:* SIP and media interfaces are assigned to the correct zones.
:* The Access Public zone routes traffic toward the Access Core zone.
:* The Access Core zone has the correct remote endpoint configured.
:* The Session Routing Policy is selected and points to the intended routing behavior.
:* Security rules only allow trusted VoIP.ms sources and trusted core-side devices.
:* CAC limits match the licensed and intended concurrent call capacity.
:* Transcoding is configured only when required.

Catelaya - Orchid Link SBC

2026-06-23T18:32:26Z

Scomtois: moved Catelaya - Orchid Link SBC to Catalaya - Orchid Link SBC

#REDIRECT [[Catalaya - Orchid Link SBC]]

Cataleya - Orchid Link SBC

2026-06-23T18:32:26Z

Scomtois: moved Catelaya - Orchid Link SBC to Catalaya - Orchid Link SBC

Cataleya - Orchid Link SBC

2026-06-23T18:28:39Z

Scomtois: Created page with "none This article provides general configuration guidelines for connecting an '''Orchid Link Session Border Controller (SBC)''' with [[VoI..."

File:Orchid Link Session Routing Policy.png

2026-06-23T13:44:36Z

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File:Orchid Link CAC Profile Zone.png

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