Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

2018-05-13 Thread Alex Balashov 
That seems correct and sounds like it should work. What is the request
URI of the INVITE that comes back into Kamailio on the B leg? Does its
domain match the AOR domain? Do you have your registrar/usrloc set to
use domains on lookup()?

-- Alex

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

2018-05-13 Thread Wilkins, Steve 
Yes, that is what has me a little boggled.  If I don't, calls do not go through 
so I am obviously confused, but that is how I was finally able to get full 
duplex calls working.

In Asterisk, I have a Kamailio endpoint listening for traffic from Kamailio. 
How would Asterisk know the Contact information if Asterisk does not re-contact 
Kamailio for that Information (since all the registration information is on 
Kamailio).
 
I was under the impression that it is best to keep the registrations on 
Kamailio, why forward them.  Below is my pjsip.conf.
When a user calls 30001@192.21.1.5 (Kamailio IP), the call is registered in 
Kamailio, and Asterisk can contact Kamailio to get the 
Contact information it needs.  


Pjsip.conf

[kamailio](!)
type=endpoint
context=from-internal
transport=transport-tcp
media_address=100.20.30.125 //Asterisk PBX IP
...
aors=kamailio

[kamailio](kamailio)
aors=kamailio

[kamailio]
type=aor
contact=sip:192.21.1.5:5060

[kamailio]
type=identify  ; Must be of type identify (default: "")
endpoint=kamailio
match=192.21.1.5

[30001](webrtc) //tls endpoint
auth=auth30001
aors=30001

[auth30001](auth-userpass)
password=12345
username=30001

[30001](aor-single-reg)
contact=sip:30001@192.21.1.5:5060

Am I totally wrong, even though it works?

Thank you all very much!
-Steve




-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex 
Balashov
Sent: Sunday, May 13, 2018 9:19 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of 
an "INVITE"

On Sun, May 13, 2018 at 01:14:08PM +, Wilkins, Steve wrote:

> In my configuration I have to use Asterisk as my PBX, and I use 
> Kamailio in front of Asterisk accepting and inspecting calls.  I have 
> many AORs, for which a phone can register to.  I have noticed that, 
> depending on the call, when a call arrives it gets REGISTERED in 
> Kamailio (I do not forward the registration to Asterisk); Kamailio 
> sends the INVITE to Asterisk, and then Asterisk send the INVITE back 
> to Kamailio.  The calls seem to work fine (Duplex Audi/Video).  I 
> think I know why the INVITE is forwarded to Kamailio but have not been 
> able to work around it; the AOR's all have a Contact of Kamailio and I 
> think this is the reason for the forwarding of the INVITE.  If I don't 
> set Kamailio as the Contact for the AOR's, calls do not work.

Are you sure that you are using the word AOR correctly? 

If you're not forwarding registrations to Asterisk but instead storing them in 
Kamailio's registrar, setting the Contact binding of an AOR to have the address 
of that very same Kamailio registrar would make no sense.

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

2018-05-13 Thread Alex Balashov 
On Sun, May 13, 2018 at 01:14:08PM +, Wilkins, Steve wrote:

> In my configuration I have to use Asterisk as my PBX, and I use
> Kamailio in front of Asterisk accepting and inspecting calls.  I have
> many AORs, for which a phone can register to.  I have noticed that,
> depending on the call, when a call arrives it gets REGISTERED in
> Kamailio (I do not forward the registration to Asterisk); Kamailio
> sends the INVITE to Asterisk, and then Asterisk send the INVITE back
> to Kamailio.  The calls seem to work fine (Duplex Audi/Video).  I
> think I know why the INVITE is forwarded to Kamailio but have not been
> able to work around it; the AOR's all have a Contact of Kamailio and I
> think this is the reason for the forwarding of the INVITE.  If I don't
> set Kamailio as the Contact for the AOR's, calls do not work.

Are you sure that you are using the word AOR correctly? 

If you're not forwarding registrations to Asterisk but instead storing
them in Kamailio's registrar, setting the Contact binding of an AOR to
have the address of that very same Kamailio registrar would make no
sense.

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

2018-05-13 Thread Wilkins, Steve 
Hello Pan,

Thank you for responding!,

In my configuration I have to use Asterisk as my PBX, and I use Kamailio in 
front of Asterisk accepting and inspecting calls.
I have many AORs, for which a phone can register to.  I have noticed that, 
depending on the call, when a call arrives it gets REGISTERED in
Kamailio (I do not forward the registration to Asterisk); Kamailio sends the 
INVITE to Asterisk, and then Asterisk send the INVITE back to
Kamailio.  The calls seem to work fine (Duplex Audi/Video).  I think I know why 
the INVITE is forwarded to Kamailio but have not been able
to work around it; the AOR's all have a Contact of Kamailio and I think this is 
the reason for the forwarding of the INVITE.  If I don't set Kamailio as the 
Contact for the AOR's, calls do not work.

Thanks Again,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, May 11, 2018 10:20 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of 
an "INVITE"

Hello Steve.

What are you trying to achieve?

The call could go from client A to Kamailio to client B. No need to involve 
Asterisk. If you need PBX functionality, the INVITE needs to be routed to 
Asterisk, which will most likely answer the call and then set up a new call to 
client B. As Asterisk doesn't know where client B is, it needs to route this 
new call to Kamailio where client B is registered. It's possible for Asterisk 
to know where client B is but that solves nothing and may create other problems.

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users 
>
 On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List 
>
Subject: [SR-Users] Question about using Kamailio and Asterisk and flow of an 
"INVITE"

Hello All,

I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using 
Kamailio and Asterisk.

Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 
from WebRTC Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the 
Invite back to Kamailio.  Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is 
registered in Kamailio for the called number.
Does this seem correct?  It seems like there is an extra hop in there.

The reason I am now very curious now is because everything works fine if using 
Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.

I would appreciate any thoughts on this.

Thank you in advance.



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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

2018-05-11 Thread Pan Christensen 
Hello Steve.

What are you trying to achieve?

The call could go from client A to Kamailio to client B. No need to involve 
Asterisk. If you need PBX functionality, the INVITE needs to be routed to 
Asterisk, which will most likely answer the call and then set up a new call to 
client B. As Asterisk doesn't know where client B is, it needs to route this 
new call to Kamailio where client B is registered. It's possible for Asterisk 
to know where client B is but that solves nothing and may create other problems.

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users  On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Question about using Kamailio and Asterisk and flow of an 
"INVITE"

Hello All,

I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using 
Kamailio and Asterisk.

Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 
from WebRTC Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the 
Invite back to Kamailio.  Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is 
registered in Kamailio for the called number.
Does this seem correct?  It seems like there is an extra hop in there.

The reason I am now very curious now is because everything works fine if using 
Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.

I would appreciate any thoughts on this.

Thank you in advance.



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