SIP Trunk with Asterisk

[KTarun003]

Hi,

I would like my agent to accept inbound calls or make outbound calls through asterisk server. I have searched the entire documentation, but I could only find the steps Twilio or Telnyx. Any help will be much appreciated.

[dennwc]

Hi @KTarun003!

The steps to accept inbound or make outbound calls for Asterisk is the same as with Twilio or Telnyx. We do not provide documentation for it, because Asterisk can be configured in many different ways and we obviously cannot cover them all.

Is there any specific step of the quickstart that doesn't work with your Asterisk setup? Also, please consider joining our Slack - we usually answer general support questions there.

[KTarun003]

I am facing issue when I want to create a trunk, I dont know which URI to point the trunk. And I am using examples/voice-pipeline-agent/function_calling_weather.py example. Are there any changes to this code when using SIP. I am trying to self-host. if that helps.

[arthurblake]

I have a similar question. I just asked it on the #sip slack channel, and I referenced this issue too.

[rodrigoGA]

I’m trying to connect it with an Asterisk system, but for now, I’m starting with a softphone like Zoiper to test the integration, and I haven’t been able to get it working correctly.
I’m using the LiveKit API to create rules similar to these:
SIP Trunk Creation: lk sip inbound create inbound-trunk.json

  "trunk": {
    "name": "Softphone inbound trunk",
    "numbers": ["+9999999999"], 
    "auth_username": "sip-user",
    "auth_password": "sip-pass",
    "allowed_addresses": ["0.0.0.0/0"]
  }
}

Dispatch Rule Creation: lk sip dispatch create dispatch-rule.json

{
  "name": "test-1 dispatch rule softphone",
  "trunk_ids": ["<id>"],
  "rule": {
    "dispatchRuleIndividual": {
      "roomPrefix": "call"
    }
  }
}

The issue is that I can’t register the softphone with LiveKit. I’ve followed the official documentation and successfully set up an example with Twilio by associating a phone number, but now I’d like to test it with Zoiper to eventually integrate the system with Asterisk.

I’m registering the softphone with the following details:

Username: sip-user
Password: sip-pass
Sip uri: 6o3m8e2088l.sip.livekit.cloud

Do you have any advice or additional steps to make this configuration work correctly?
Any help would be greatly appreciated!

[charly17]

Hi,
In my case, I successfully completed the integration with Asterisk with the help of fellow developers and by using LiveKit documentation. The process involves creating a dispatch rule and a room to obtain the SIP URI and registering the number within LiveKit.

You should not register the trunk outside of LiveKit for outgoing calls. For incoming calls, you only need to send the call to the SIP URI you get from LiveKit.

On the server where you have Asterisk and currently receive the DID from your provider to a standard Asterisk extension (this is to verify that everything is working fine on your provider's side and with your Asterisk server), once validated, you should route the call from your Dialplan like this:
same => n,Dial(SIP/${EXTEN}@dilivekitetcetc.sip.livekit.cloud)

As mentioned, the call must negotiate using the ulaw-alaw codecs, as I encountered issues when only G729 was being sent in my configuration.

For outgoing calls, you need to configure LiveKit so that when the server launches, it handles the call to the destination number (this is the "participant" according to LiveKit documentation). You will also need to create a trunk in Asterisk and register it in LiveKit since it will generate the call to your SIP provider connected to the PSTN. Therefore, you must configure Asterisk to manage this process (create the context where the call will land and send it to your SIP provider).

[dydimos]

Hi,
I try to get livekit also running with asterisk, but I have a complete local setup with an ATA to connect an analog phone, livekit and livekit-sip locally running in a docker container. I managed it that when I call a defined number, the phone enters the room (I use "dispatchRuleDirect"). But the phone continues to ring - the room is not answering the phone.

Any help would be appreciated - especially @charly17 - do you maybe have an idea whats the problem on my side?
Thanks in advance :)

[dennwc]

I think one way to resolve the issues here would be if someone can provide an example of Asterisk configuration inside a Docker container.

We can then combine it with our existing Docker compose example for self-hosting and properly document integration with Asterisk in our official docs.

Unfortunately I don't have spare cycles right now to do it, and I'm not very familiar with Asterisk. Can someone help by making a repo/PR with an example config? That would be really helpful!

[dtcgroup]

Hi, I try to get livekit also running with asterisk, but I have a complete local setup with an ATA to connect an analog phone, livekit and livekit-sip locally running in a docker container. I managed it that when I call a defined number, the phone enters the room (I use "dispatchRuleDirect"). But the phone continues to ring - the room is not answering the phone.

Any help would be appreciated - especially @charly17 - do you maybe have an idea whats the problem on my side? Thanks in advance :)

I know it is not an answer, but I am having exactly the same problem using FreeSwitch.

It sends the Invite, gets a Processing(100) response and then a Ringing (180) response repeated every second until the call times out with no answer.

My node application detects the call and logs the connecting to agent which suggests the dispatch and numbers all line up. We just never get the call answered (Status 200) by the room as you say.

My current line of thought is if the response is coming from a different IP address so is being blocked by the firewall...

It is similar to : #208

Update: I added a pin to the dispatch rule. This stopped the agent from doing anything but the call did the same thing. This suggests LiveKit cloud thinks the call is being answered but the return SIP messaging is going to the wrong place.

Update 2: It works where the SIP messaging is TCP but not when it is UDP. To set it up on FreeSwitch is really easy.

Create a Destination setting the action as a bridge to the SIP endpoint like:

<extension name="TestAgent" continue="false" uuid="4f38bc1b-fa5d-44ef-988c-c6191432ab19">
	<condition field="destination_number" expression="111">
		<action application="bridge" data="sofia/external/441234123456@zzzzzzzzzz.sip.livekit.cloud;transport=tcp"/>
	</condition>
</extension>

You don't need to do anything else. No need to create a trunk.

[dennwc]

If you get continuous 180 responses, this means LiveKit SIP is waiting for any media from an Agent.

Due to the way SFU works currently, publishing tracks is not enough. You might need to send at least one media packet (could be empty/silence). This should unblock SIP and it will start bridging the call.

Are you seeing Waiting for track subscription(s) each time the call gets stuck? If so, this is it.

[dydimos]

Somehow I was able to solve it, now the call is transmitted to the room and I can hear myself speak. I used your docker-compose as a reference.

If everything works for me, I will document the setup and can post it here if there is any interest.

[umairmehar]

Hello @dydimos, I'm trying to achieve the same.

I have a complete local setup with an ATA (Grandstream H802) to connect an analog phone, livekit, and livekit-sip locally running.

The ATA modem is simply not registering/communicating with the local livekit-sip server.

I confirmed that all my servers, ATA modem, livekit, livekit-sip, and Redis, are running and accessible on the local network.

Any help would be appreciated.

Thanks in advance.

[duzaq]

Estou tentando conectá-lo a um sistema Asterisk, mas, por enquanto, estou começando com um softphone como o Zoiper para testar a integração, e não consegui fazê-lo funcionar corretamente. Estou usando a API do LiveKit para criar regras semelhantes a estas: Criação de tronco SIP: lk sip inbound create inbound-trunk.json

  "trunk": {
    "name": "Softphone inbound trunk",
    "numbers": ["+9999999999"], 
    "auth_username": "sip-user",
    "auth_password": "sip-pass",
    "allowed_addresses": ["0.0.0.0/0"]
  }
}

Criação de regra de despacho: lk sip dispatch create dispatch-rule.json

{
  "name": "test-1 dispatch rule softphone",
  "trunk_ids": ["<id>"],
  "rule": {
    "dispatchRuleIndividual": {
      "roomPrefix": "call"
    }
  }
}

O problema é que não consigo registrar o softphone com o LiveKit. Eu segui a documentação oficial e configurei com sucesso um exemplo com o Twilio associando um número de telefone, mas agora eu gostaria de testá-lo com o Zoiper para eventualmente integrar o sistema com o Asterisk.

Estou registrando o softphone com os seguintes detalhes:

Nome de usuário: sip-user Senha: sip-pass Sip uri: 6o3m8e2088l.sip.livekit.cloud

Você tem algum conselho ou etapas adicionais para fazer essa configuração funcionar corretamente? Qualquer ajuda seria muito apreciada!

Alguma solução?

[UlfWil]

Somehow I was able to solve it, now the call is transmitted to the room and I can hear myself speak. I used your docker-compose as a reference.

If everything works for me, I will document the setup and can post it here if there is any interest.

Of course, I'm interested in seeing your setup. Do you use livekit.cloud, or are you running a local LiveKit server?

[UmairShah7677]

Hi everyone,

I'm attempting to integrate my Asterisk server with LiveKit SIP (Also mentioned this in slack community), and I'm running into an issue where calls to the SIP URI are failing with a 503 "Try again later" response. Below are the details of what I've done so far:

Setup Overview:

• SIP URI: sip:2vbic7bb4fd.sip.livekit.cloud
• Asterisk is configured to forward calls to this URI when the DID is dialed.
• The DID is confirmed to be working correctly and is properly routed to the context in Asterisk.

Asterisk Dialplan:

Here’s the relevant part of my dialplan:

exten => 9999,1,Dial(SIP/2vbic7bb4fd.sip.livekit.cloud)
exten => 9999,n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?retry:hangup)
exten => 9999,n(retry),Wait(5)
exten => 9999,n,Dial(SIP/2vbic7bb4fd.sip.livekit.cloud)
exten => 9999,n(hangup),Hangup()

Issue:

When I call the DID, Asterisk attempts to route the call to LiveKit, but I receive the following logs:

-- Called SIP/2vbic7bb4fd.sip.livekit.cloud
-- Got SIP response 503 "Try again later" back from 84.235.238.224:5060
-- SIP/2vbic7bb4fd.sip.livekit.cloud-0000002d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

This results in a CONGESTION status and retries as per the dialplan logic. However, the issue persists even after retries.

Steps Taken to Troubleshoot:

1. Codec Compatibility: I ensured that only ulaw and alaw codecs are allowed in both Asterisk and LiveKit.
2. SIP URI Verification: Confirmed the SIP URI is correct and matches the one provided by LiveKit.
3. Firewall: Verified that there are no firewall restrictions blocking traffic to/from LiveKit (84.235.238.224:5060).
4. LiveKit Dispatch Rules: Checked that the dispatch rule routes calls to the correct room in LiveKit.
5. Connectivity: Tested connectivity to the LiveKit server using ping and telnet for SIP port 5060.

Logs from Asterisk:

Here’s an excerpt from the logs:

Got SIP response 503 "Try again later" back from 84.235.238.224:5060
SIP/2vbic7bb4fd.sip.livekit.cloud-0000002d is circuit-busy

Request for Help:

• Is the 503 "Try again later" response indicative of an issue on LiveKit's side, or could it be related to my configuration?
• Are there additional debugging steps I can take to identify the cause of this issue?
• Could this be related to resource availability or misconfigured dispatch rules in LiveKit?

Any guidance or suggestions would be greatly appreciated. Thank you in advance for your time and help!

Best regards.

[MefhigosetH]

Hi @UmairShah7677,

I don't see that Asterisk is sending the extension number when Dial. You can try this: Dial(SIP/EXT_NUMBER@2vbic7bb4fd.sip.livekit.cloud) where EXT_NUMBER is the number that you configured in Livekit Cloud when created the Inbound Trunk.

Remember that you need configure two things in Livekit: an Inbound Trunk, and a Dispatch Rule. As documentation says: https://docs.livekit.io/sip/trunk-inbound/

[ccameroVzy]

Hi everyone!

We are having a very similar issue.

We are using the cloud sip server, and first we connected our agent using sip-trunk and dispatch rules to a Twilio Number and it works just fine.

But we need to integrate with Kazoo our VoIP, so we created an extension on kazoo to LiveKit, created the SIP Trunk and dispatch rules. But in this case when we make the call, its received by LiveKit, the agent is actually dispatched to the room and we can see even the agents starts the speech

2025-01-20 14:14:05,947 - DEBUG livekit.agents.pipeline - speech playout started {"speech_id": "01fa05decd9a"}

but on the user ends theres still ringing, and we just keep getting a 180 code and never de 200 status

Any guidance is helpful

[MefhigosetH]

Hi @ccameroVzy,

I've read that Livekit Cloud, only works with TCP packets. Not UDP. Try with TCP communication.

[ccameroVzy]

@MefhigosetH yes, that was the issue.

Now is working properly thanks!

[UmairShah7677]

Hey @MefhigosetH Thank you for help, it is connected but now it is keep ringing. Can you please tell me @ccameroVzy that how you fixed as after this Dial(SIP/EXT_NUMBER@2vbic7bb4fd.sip.livekit.cloud)
the 503 error is solved, but its keep on ringing, Also i can see session on the livekit cloud.

In sngrep it is giving response of "200" but i can't hear any agent.

[MefhigosetH]

@UmairShah7677 , I think that your Asterisk needs to communicate with Livekit Cloud via TCP. To do so, follow this article: https://support.digium.com/s/article/How-to-use-SIP-over-TCP

Note that Asterisk 1.8 and above support SIP over TCP.

If upgrade is not possible, you can try a hosted Livekit instance using Docker Compose. This schema support standard SIP via UDP. I have been test with Asterisk 11 and everything works fine.

[UmairShah7677]

Thank you @MefhigosetH working fine.