Notify using SIP server #3781

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opened 2026-02-28 03:40:45 -05:00 by deekerman · 8 comments
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Originally created by @Hoto-Cocoa on GitHub (Nov 24, 2024).

I searched about SIP but there was exist only about monitoring SIP server but not using SIP server.

🏷️ Feature Request Type

New notification-provider

🔖 Feature description

I have local SIP server and I'm using this as my emergency contact.

Now I want integrate my Uptime Kuma instance to my SIP server.

✔️ Solution

Add SIP server to notification method. If monitor goes down, Dial to specified number.

I think just dialing only is OK, but If I can play audio file or Uptime Kuma can say sentences will be awesome.

Alternatives

No response

📝 Additional Context

No response

Originally created by @Hoto-Cocoa on GitHub (Nov 24, 2024). ### 📑 I have found these related issues/pull requests I searched about SIP but there was exist only about monitoring SIP server but not using SIP server. ### 🏷️ Feature Request Type New notification-provider ### 🔖 Feature description I have local SIP server and I'm using this as my emergency contact. Now I want integrate my Uptime Kuma instance to my SIP server. ### ✔️ Solution Add SIP server to notification method. If monitor goes down, Dial to specified number. I think just dialing only is OK, but If I can play audio file or Uptime Kuma can say sentences will be awesome. ### ❓ Alternatives _No response_ ### 📝 Additional Context _No response_
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@CommanderStorm commented on GitHub (Nov 25, 2024):

This is kind of niche/uncommon notification provider.
If you/someone else would like to contribute such a provider, here is the Contribution guide:
https://github.com/louislam/uptime-kuma/blob/master/CONTRIBUTING.md

If you/someone else do decide to do this, please not these two suggestions, which often push PRs back by a review-cycle:

  • Ensure that the PR includes screenshots of the DOWN/UP and testing/cert-expiry events.
  • All keys that can/should be translated are in en.json (⇒ our translators can translate this and no merge-confilicts between weblate and master can arise)

Related Notification provider-requests:

@CommanderStorm commented on GitHub (Nov 25, 2024): This is kind of niche/uncommon notification provider. If you/someone else would like to contribute such a provider, here is the Contribution guide: https://github.com/louislam/uptime-kuma/blob/master/CONTRIBUTING.md If you/someone else do decide to do this, please not these two suggestions, which often push PRs back by a review-cycle: - Ensure that the PR includes screenshots of the `DOWN`/`UP` and `testing`/`cert-expiry` events. - All keys that can/should be translated are in `en.json` (⇒ our translators can translate this and no merge-confilicts between weblate and master can arise) Related Notification provider-requests: - #3295
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@kristiandg commented on GitHub (Nov 26, 2024):

I've asked my developers to submit a pull request for this. I asked for a similar module a couple years ago. This module can register as a SIP endpoint or just do an OPTIONS request. We didn't build it for a NOTIFY because that's an uncommon method for SIP health, but here's the pull request, which I'm hoping gets approved. https://github.com/louislam/uptime-kuma/pull/5382

@kristiandg commented on GitHub (Nov 26, 2024): I've asked my developers to submit a pull request for this. I asked for a similar module a couple years ago. This module can register as a SIP endpoint or just do an OPTIONS request. We didn't build it for a NOTIFY because that's an uncommon method for SIP health, but here's the pull request, which I'm hoping gets approved. https://github.com/louislam/uptime-kuma/pull/5382
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@Hoto-Cocoa commented on GitHub (Nov 26, 2024):

Anyways then I will try implement it myself when I have time for this.

I think I don't have any time for do this now, but I hope I have more time to do this later.

@Hoto-Cocoa commented on GitHub (Nov 26, 2024): Anyways then I will try implement it myself when I have time for this. I think I don't have any time for do this now, but I hope I have more time to do this later.
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@kristiandg commented on GitHub (Nov 29, 2024):

@Hoto-Cocoa, what is your SIP server? Is it something like Asterisk, or something like NetSapiens? I ask because, something like NetSapiens, has the ability to execute web instructions to tell the SIP server to do things. There's an entire structure utilizing something called a WebResponder that allows you to instruct the server to speak via it's native TTS engine (or, of course, play a sound file).

If you're using something like FreePBX, then you could use the Voicemail Notification module (which calls out to an escalating set of numbers, or just the one number), whenever a voicemail is left, and you could generate a phone call to leave a voicemail that would then trigger that event.

@kristiandg commented on GitHub (Nov 29, 2024): @Hoto-Cocoa, what is your SIP server? Is it something like Asterisk, or something like NetSapiens? I ask because, something like NetSapiens, has the ability to execute web instructions to tell the SIP server to do things. There's an entire structure utilizing something called a WebResponder that allows you to instruct the server to speak via it's native TTS engine (or, of course, play a sound file). If you're using something like FreePBX, then you could use the Voicemail Notification module (which calls out to an escalating set of numbers, or just the one number), whenever a voicemail is left, and you could generate a phone call to leave a voicemail that would then trigger that event.
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@Hoto-Cocoa commented on GitHub (Nov 29, 2024):

@kristiandg I'm currently using Asterisk only.

@Hoto-Cocoa commented on GitHub (Nov 29, 2024): @kristiandg I'm currently using Asterisk only.
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@kristiandg commented on GitHub (Nov 29, 2024):

@Hoto-Cocoa OK, so raw Asterisk and not the FreePBX variant. Just wanted to make sure. Another thing you could do is register an ATA to it, and set that to auto call an extension or queue in Asterisk (and in that extension or queue, you have it set to redirect to the numbers you want called). Then, all you'd need is a contact closure to short out the analog port on the ATA, which will then generate the phone call. That's a very "analog" way to achieve it, but in a pinch, I just wanted to mention it as well. :)

@kristiandg commented on GitHub (Nov 29, 2024): @Hoto-Cocoa OK, so raw Asterisk and not the FreePBX variant. Just wanted to make sure. Another thing you could do is register an ATA to it, and set that to auto call an extension or queue in Asterisk (and in that extension or queue, you have it set to redirect to the numbers you want called). Then, all you'd need is a contact closure to short out the analog port on the ATA, which will then generate the phone call. That's a very "analog" way to achieve it, but in a pinch, I just wanted to mention it as well. :)
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@cmar7945 commented on GitHub (Jul 16, 2025):

Something like this would be awesome. Being able to use something like asterisk or freepbx to send an sms when a service goes down. The current ones like clicksend and twilio are business only for sms. Can't use it outside of a biz environment which sucks.

@cmar7945 commented on GitHub (Jul 16, 2025): Something like this would be awesome. Being able to use something like asterisk or freepbx to send an sms when a service goes down. The current ones like clicksend and twilio are business only for sms. Can't use it outside of a biz environment which sucks.
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@officialh1 commented on GitHub (Sep 30, 2025):

This is useful for Public Trunking. Some companies trunk to the cloud and provide their own SIP services to services like Teams or get their service from smaller providers or even big ones like Verizon. Although some are going all cloud, many prefer on-prem still to this day.

@officialh1 commented on GitHub (Sep 30, 2025): This is useful for Public Trunking. Some companies trunk to the cloud and provide their own SIP services to services like Teams or get their service from smaller providers or even big ones like Verizon. Although some are going all cloud, many prefer on-prem still to this day.
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