[asterisk-users] what is a SIP invite, and who can issue them?

[Thufir]

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."

http://www.3cx.com/blog/voip-howto/sip-invite-header-fields/


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
<https://portal.telnyx.com/#/app/telephone-data> Tab

Send a SIP Invite to *lrnlookup.telnyx.com <http://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
172.16.1.12;branch=z9hG4bKfae8cb69f547b8cb;received=172.16.0.179
From: <sip:5555555555 at 172.16.1.12>;tag=102
To: <sip:5555555555 at 174.36.199.131>
call-id: 0704037283648236478326200101 at 172.16.1.12
CSeq: 1 INVITE
Contact: Transfer <sip:5555555555;*rn=+15555555556;npdi;*@174.36.199.131>
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.


thanks,

Thufir
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