Hi, I’m interested in subscribing to your phone system and would like to confirm whether it supports integration with a third-party SIP trunk provider.
Thanks,
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Hi @Odyssey,
Yes, it does. You’ll need to ask us to enable it as it’s not available by default.
More info here:
Great, thanks for the clarification!
I’d like to confirm:
-
Is your phone system license charged per user and/or per device?
-
With one user account, how many devices can be used simultaneously?
Looking forward to your response!
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It’s per phone system instance - so you can add multiple “users”
There is a soft limit of 30 devices however if you need more you can let us know.
We have two numbers (from a third party)—one for the office and one for a representative who is located in another country and frequently moves. I would like to know if it’s possible to set up these two numbers in your phone system, with each number assigned to the respective user as described.
Additionally, to clarify, would we need two separate SIP trunks, one for each number?
I also want to ensure that the setup for these two users will be separate, and that their calls will not ring interchangeably or cause any conflicts.
Thank you for your assistance. I look forward to your response.
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Yes, that can easily be done with the phone system. You can set up a SIP account for each and have independent routing for each.
You can even set up multiple workspace tabs to keep the logic completely separate.
Hi,
I already have the phone system set up. Could you please enable the Trunk feature for me?
Thank you in advance!
Hi,
I have already configured the Inbound Trunk. Could you please confirm if I also need to create a call flow in order to receive calls from the SIP Trunk? I added the Phone number, and created a Call Flow, but when I try to call the number from the trunk provider, it says the number is unavailable. What could be the issue?
After configuring the Inbound Trunk, I noticed that the SIP Trunk status in my trunk provider’s portal showed as “Online” for a short time, but then it turned “Offline” and I have not been able to bring it back online since. I have to delete the trunk and add it again to bring it back online, but I’m afraid it will turn offline again. Could you please help me investigate this?
I understand that the trunk must remain consistently online in order to send and receive call signals. Could this issue be related to server expiry time, registration interval, or refresh settings? If so, could you provide instructions or recommended values for these settings to help keep the trunk registered and stable?
Additionally, I’ve added a phone number from the trunk provider into the PBX and created an Outbound Trunk. However, the number does not appear as an available option in my softphone application. Is there an additional step required to associate the number or extension with the softphone?
Thank you in advance for your assistance.
Thanks for the detailed information about your trunk setup issues. I’ve consulted with our technical team about your situation.
Our team has investigated and found that no INVITE signals are reaching our inbound trunk with your domain. This suggests that calls from 61290982328 aren’t being properly forwarded to your trunk endpoint (XXXX.in.phone.systems) or there might be a configuration error.
Here are some recommended steps to resolve this:
- Check your DID provider configuration:
- Could you share screenshots of how you’ve configured the trunk on your provider’s side?
- Specifically, check if they’re using IP authentication only
- Registration settings:
- Try setting the registration timer to 3600 seconds (standard value)
- If your provider only supports IP authentication, you might need to remove the registration from your current Inbound trunk at phone.systems and set up the trunk on the provider’s side to send INVITE to the configured SIP URI
- For your outbound number issue:
- Your third-party number should be added in the Phone Number section
- Then select it as CLI for outbound calls under Contact Methods
- You can find this at: Contact Methods Documentation
- Look specifically under “Outbound calls → Enable external outbound calls” for the Caller IDs section
It might be worth asking your DID provider for a pcap (packet capture) when a call is initiated toward your DID and then forwarded to the given SIP URI. This would help diagnose exactly where the communication is breaking down.
Let me know if you need any clarification and/or if this persists.
Hi,
In previous message, I’m experiencing an issue with the SIP Registration mode, so I switched to IP Authentication mode, which seems to be working. However, in IP Auth mode, I don’t have a username and password to register the Termination Gateways (Outbound Trunk).
Although I’m able to make outgoing calls using the third-party number, these calls are not appearing in the third-party’s outbound call reports. It seems there may be a configuration issue with either the Outbound Trunk settings or the Outbound Rules.
Could you please guide me on how to correctly configure the Outbound Trunk or Outbound Rules to ensure proper call logging?
Thanks,
For IP Authentication mode, you typically don’t need username/password credentials in the Termination Gateway configuration. Instead, your trunk provider authenticates based on your IP address. In the Termination Gateway settings, you can leave the Username and Password fields blank and just configure:
- Host: Your provider’s SIP server
- Port: Usually 5060 or 5061
- Network Protocol: UDP or TCP (as required by your provider)
This issue is likely related to the CLI (Caller ID) being sent to your provider. Check your Termination Route configuration:
- Termination Route Settings: Make sure your route is properly configured with the correct Gateway assigned
- CLI Rules: In your Termination Route, check the SRC Rewrite Rule. You may need to ensure the source number being sent matches the format your provider expects for your assigned number
- Number Format: Verify that the CLI being sent matches exactly what your provider has assigned to you (with or without country codes/prefixes)
If your provider expects calls to originate from a specific number format, you might need to set up a SRC Rewrite Rule in your Termination Route to ensure the correct CLI is sent.
Can you check what CLI/source number is being sent in your outbound calls and confirm it matches what your third-party provider has on file for your account?
Hi,
I’m following up on the issue I raised regarding the connection with the third-party SIP trunk. I previously sent you an email along with the PCAP files from both the third-party side and our side, as requested.
However, I haven’t received any response yet. Could you please let me know if you’re currently working on this case? We need an update so we can continue our investigation on our end as well.
This is an urgent matter for us, as it’s part of our transition from Skype, which will be expiring soon.
As a quick update: I’m now able to make outbound calls using the third-party number through the correct third-party trunk. I can see the outbound call logs in their report, which confirms the calls are going through their system. However, I also noticed that the same outbound calls appear in your Call Analytics as well.
Could you please confirm whether your Call Analytics logs all outbound calls, even those routed through third-party providers? I just want to ensure that there are no misconfigurations in my outbound rules that might result in us being charged for the same calls by both you and the third-party provider.
Looking forward to your response.
Thanks,
Apologies on the delay with this @Odyssey, and please look out for our email shortly on this.