How to configure TATA SIP trunk (provider in india)

How to configure TATA SIP trunk (provider in india)
They are provide following information;
DID no : 7316832500
Start Range : 7316832500
End Range : 7316832589
Customer IP :
Gateway IP :
Subnet Mask :
SIP Server IP :
Username &; password is also given to me

how to setup a sip trunk with the above information, And make outbound in inbound routes.

Please help…

when select outbound then only username and auth username is activate other wise not activating cen give some explanation

Your customer IP address, the gateway and the netmask go in network settings. Note that this is from an old posting, so may not match your system, and might possibly be commercial, in which case you may need to do this directly on Linux:

You will need static, or to configure a DHCP rule to set the address.

Your DIDs go in inbound routes. Exactly how you do this depends on how you want to use them. You should include matches for all 90, although some may be wildcard.

Please note I’m doing this off documentation, as my expertise is more on Asterisk than FreePBX

@david55 i think you did post at Wrong channel or Topic mate :slight_smile:

I’m telling you how to make use of Customer IP, gateway, netmask, and the DIDs in the information with which you were provided, none of which go on the page you showed. I don’t see how I’m on the wrong topic. I can’t tell you exactly how to use the DIDs, as that is a local policy decision.

@david55 you are telling me or someone else… Because i have not asked anything for DID :slight_smile:

DID is part of the above information. and, although it doesn’t use the term, these are also DIDs (90 of them):

after add network settings what i do
that wired sip trunk don’t have a internet is there any problem.

If you have no internet connection, you will have problem with DNS. I believe it is possible to use chan_pjsip without it, but I’m not sure of the specifics. Do you have internet via another interface, or is this a completely isolated VoIP system? If you have another interface, to the real internet, I’m not sure that you can complete the configuration without going direct to Linux, as I think the gateway, in the above form, is a default gateway, and you would need a specific gateway to the provider’s server. I don’t have the actual form to play with and it is possible more options appear when you select Static.

You’d need a Linux routing rule something like “route add -host gw” at the Linux level, if you also have a default route that does cover the internet, to stop the default route being used for the service.

You’ve got to the limits of things for which I can give specifics.

PS “directly on Asterisk”, in my first post, should be “directly on Linux”.

If you are fully isolated, and they don’t provide a working DNS environment, there may be some hints in: DNS Availability - Phones

Note that, in such an environment, you will have difficulty updating FreePBX and some commercial parts may refuse to work.

If you have a second interface, with connectivity to the real internet, the following may be helpful, although it seems to suggest that you may have to go direct to Linux: Two NICs -- One is configured with Local Lan and Second will be P2P to my Local Service Provider

when i enter static ip, that only showing fwconsole but that ip cannot be changing and also not login to the gui.

send me docment for Asterisk,i dont know the how to run the Asterisk run in linux cen os 7.

Something very strange here! Exactly the same question, right down to the customer specific parameters, was asked almost 3 years ago! I think it is word for word identical.

Also, I didn’t notice it before, but the netmask provided wouldn’t allow the operation of VoIP network using the router they provide, as there are no free addresses. The network itself is …88, the gateway is … 89, the PABX or phone is … 90, and the local broadcast address is …91. That means the only viable configurations are:

  • a single phone;
  • a two interface (possibly one with VLANs) PABX, with the second interface carrying the existing LAN and internet connection; or
  • connecting through a local router onto that existing LAN, in which case the local router is the one that needs to be configured with Tata’s network information.

I also notice the reference to Team in the salutation, which may reflect unreasonable expectations by the OP. People replying here are not an organised team, paid for by Sangoma, but ad hoc volunteers. It’s a peer support forum, so there is no guarantee of a timely answer, or even one at all.

Given the OPs level of knowledge, I’d suggest they go in baby steps. Ideally start with a hard SIP phone and get that working, with no PC at all. Next move to PC with a soft phone, and understand the networking on the Tata side. If they are a Windows person, probably start with Windows then get the same working on Linux. Only then try introducing the PABX.

I did find some information on FreePBX and Tata, but I think it may assume too much for the Op. It is for ViciDial, which is based on FreePBX, and it uses the obsolete chan_sip. However, the key point is it uses the two interface configuration, listed above, and uses routing entered directly into Linux: How to configure tata sip trunk in asterisk vicidial

when i enter static ip, that only showing fwconsole but that ip cannot be changing and also not login to the gui.

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