What Is A SIP Invite, And Who Can Issue Them?

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Asterisk Users 3 Comments

I don’t understand what a SIP invite is. Certainly it’s explained as:

“This article explains the main fields included in a SIP INVITE, which is sent to set-up a VoIP call. A SIP INVITE message contains typically between
4 and 6 header entries with contact information inside them.”


The article enumerates the headers and explains them. But what sends the invite? Asterisk? A soft-phone?

I found sample config’s to send an invite with Asterisk but no other method was given. Can only Asterisk send an invite? Why? The article says that it’s sent “to set-up a VoIP call,” so presumably any reasonable soft-phone sends these invites as a normal process.

That’s all well and good, but how do send an actual invite and get a response? This can only be done through Asterisk?

This is in the context of:

Requires IP Authentication to be setup through the portal and associated with LRN under Telephone Data

Send a SIP Invite to *lrnlookup.telnyx.com *
with the number you wish to dip on port 5060

The response will be a SIP 302 redirect for example:
SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP;branch=z9hG4bKfae8cb69f547b8cb;received2.16.0.179
From: ;tag2
call-id: 0704037283648236478326200101@
Contact: Transfer
Content-Length: 0

If a number has been ported the response will contain the dip indicator
(“npdi;”) as well as the LRN (rn=+1..), otherwise these fields will be missing

from https://apidocs.telnyx.com/
and then clicking “Data API” and then “SIP request” for details.

I have a running instance of Asterisk. I would have to handle the invite through Asterisk and keep it running to make and receive calls? Presumably this invite is interacting with Asterisk, or something similar, at telnyx.com — which seems overkill.



3 thoughts on - What Is A SIP Invite, And Who Can Issue Them?

  • Thufir wrote:

    An INVITE is a request to set up a session, commonly referred to as a call. Anything supporting SIP to establish calls uses INVITE to do so. It’s equivalent to picking up the phone and dialing a number.

    In the case of chan_sip the sip.conf file is used to configure the settings for making the call, and extensions.conf is the logic which describes when to place an outgoing call when we get an incoming call.

    Whether you must use Asterisk is really up to your experience, what you are trying to do, and whether other things can do what you need.

  • an INVITE would never be sent unless a call, or other communication with an endpoint, was being attempted?