asterisk users - Jul 2016 - what is a SIP invite, and who can issue them?

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160629/d43a6315/attachment.html>

Thufir wrote:
> 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?
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.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

On Wed, 29 Jun 2016 06:38:53 -0300, Joshua Colp 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.

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


thanks,

Thufir