Jio SIP Trunk setup

Hi Folks,

I am getting a SIP trunk installed on my premises from Jio (India).
The service provider, i.e., Jio, is saying that I would need a PBX on my premises on which they will terminate the connection.
I have set up Asterisk on a computer and I want to use that as my PBX solution.
Do I still need to have some PBX router thing for this, or as I understand, if the SIP trunk terminates in an Ethernet (RJ45) point, then I should be able to connect my computer with Asterisk running and have all its capabilities.

Please help me out here as my service provider is not clarifying this.
If I have the SIP trunk terminating in an RJ45, which i can connect either to my PC running Asterisk or another router, and then connect my PC to this router, will that work as expected?
Or, do i still need to buy some specialized hardware?
If I need a specialized hardware, what will it be?

Thanks and regards,
Mayan

you will need to know the ip address(es) and port(s) and protocol(s) of the inbound trunk(s) and the ip address(es), port, protocols and dial format for outbound calls, having that between the wiki at the top of this page and google should get you going with the voip endpoint of choice, it doesn’t need to be a PBX, it could be as little as ‘soft phones’ on pc’s or phones, an ATA for a phone or two, or as much as a full on SIP proxy. A PBX is in the middle of difficulties, the soft-phone solution the easiest to prove it is working in principle

Asterisk on it’s own will prove challenging if a noobie, but FreePBX and others products add a gooey front end to asterisk.

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Thanks for the detailed explanation. This makes sense.
Asterisk is complex, but I believe I will be able to handle the complexity.
I have a soft phone and have several laptops and mobile devices on which I can configure soft phones.
This FreePBX thing works better for US and Europe. For India I could not find.
Do you know a similar thing for India?

They seem to use a 100.64/10 private network, which typically means they provide up to an including the router. However, if it is not too late, I would look for a provider with a web site that has good input from its technical people. All I could find on their site appeared to have been written by the marketing team, so is basically almost all non-technical contact for non-technical business people.

Yeah, technical support is not great.
When you say including the router, do you mean any router, like DIR-1210/DSBNA AC1200 Wi-Fi Router | D-Link (dlink.com) ?

Or some ip pbx thing like, Uc200-30 Asterisk Mini Ip Pbx Support 30 Concurrent Calls And 120 Users With Fxo Fxs Port - Voip Products - AliExpress

What would be a better fit?
I already have an IP phone ( [Grandstream GRP2601P IP Phone]) to verify the sip line.

Also, the installation team has shared this list of pre-requisites,

Mandatory requirements:

  1. PBX should be able to send the traffic in E.164 format in To, from, and request URI
  2. Customer gateway should support SIP protocol
  3. Customer Voice Gateway Supports GE Electrical interface for connecting to JIO L2
    switch
  4. Gateway should support configuration of IP (given by JIO) for WAN SIP link
  5. PBX should be able to support PRACK (Provisional Acknowledgement)
  6. PBX should support OPTION’s messages
  7. PBX Should be able to support RFC2833 DTMF method, payload can vary
  8. PBX Should support Timer
  9. PBX should have G.711A/U codec support
  10. Customer is responsible for configuration, provisioning of Voice Gateway with
    VLANs, Host IP, Gateway IP, Subnet, DID, Pilot range given by JIO
  11. Customer is expected to test out all the options (incoming , outgoing, Different
    Circle calling across operators) in a weeks’ time from Delivery of SIP services by JIO
    Nice to have
  12. PBX supporting Update message call flow
  13. PBX sending G.711A CODEC only
  14. PBX Set for refresh update after 1800 sec.
  15. PBX supporting RTCP
  16. PBX sending Q.850 signaling in BYE message / error response
  17. PBX supporting Media attributes for HOLD/RESUME conditions.
  18. PBX should not sending
  19. PBX sending 18X message with SDP ( Ring back from PBX)
  20. PBX supporting delayed offer method ( INVITE without SDP )

While I am thinking that if I can get a RJ45 then I should be able to setup Asterisk.

Please help @dicko @david55

I think I mean JIO L2 switch and I was saying they provide that, on your site. However, I really need a network diagram. Their interface seems to Gigabit Ethernet, so, to be on the safe side, assume that you have to provide the cable and male connectors.

I think the Gateway can double with the PBX.

Item 18 seems to be incomplete.

RJ45 is loose and incorrect terminology. You want an 8P8C connector wired for 1Gb Ethernet use. and connected to a 1Gb Ethernet interface. As it isn’t totally clear, have a crossover cable available, as well.

I think FreePBX can do 19, but I’m only certain for Asterisk.

I think chan_pjsip does 16, but I’m less sure about chan_sip, and you should resist any attempt for them to provide you with a chan_sip configuration.

19 and 20 are mutually incompatible. Sending early media from the callee requires that you know the media address for the caller, and that is not available until after answer when using delayed offer. They may just be hedging their bets, by allowing either.

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