VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
hawat.thufir at gmail.com
Guest
|
 Posted: Wed Jun 29, 2016 3:47 am Post subject: [asterisk-users] what is a SIP invite, and who can issue the |
 |
|
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 Tab
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 172.16.1.12;branch=z9hG4bKfae8cb69f547b8cb;received=172.16.0.179
From: <sip:5555555555@172.16.1.12 ([email]sip%3A5555555555@172.16.1.12[/email])>;tag=102
To: <sip:5555555555@174.36.199.131 ([email]sip%3A5555555555@174.36.199.131[/email])>
call-id: 0704037283648236478326200101@172.16.1.12 (0704037283648236478326200101@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 |
|
| Back to top |
|
 |
jcolp at digium.com
Guest
|
| Posted: Wed Jun 29, 2016 4:39 am Post subject: [asterisk-users] what is a SIP invite, and who can issue the |
|
|
Thufir wrote:
| Quote: | 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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
hawat.thufir at gmail.com
Guest
|
| Posted: Wed Jul 06, 2016 3:12 am Post subject: [asterisk-users] what is a SIP invite, and who can issue the |
|
|
On Wed, 29 Jun 2016 06:38:53 -0300, Joshua Colp wrote:
| Quote: | 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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
jcolp at digium.com
Guest
|
| Posted: Wed Jul 06, 2016 4:14 am Post subject: [asterisk-users] what is a SIP invite, and who can issue the |
|
|
thufir wrote:
| Quote: | On Wed, 29 Jun 2016 06:38:53 -0300, Joshua Colp wrote:
| Quote: | 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?
|
Yes.
--
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
