Re: [SR-Users] Kamailio / Asterisk and multidomain

2017-08-22 Thread Cyrille Demaret 
I feel stupid but indeed setting up Kamelio as proxy in Asterisk did the trick.

Regards,

Cyrille


De : sr-users [mailto:sr-users-boun...@lists.kamailio.org] De la part de 
Vladyslav Zakhozhai
Envoyé : mardi 22 août 2017 15:10

Hi,

I think that this issue is more specific for asterisk than kamailio.

Maybe configuration of outbound proxy may help here but I am not sure.

On Tue, Aug 22, 2017 at 3:50 PM, Cyrille Demaret  
wrote:
Hi,

I'm new to Kamailio and I'm trying to use it as a front SIP proxy to one or
more asterisk. Unlike to the "Kamailio 4.0.x and Asterisk 11.3.0 Realtime
Integration" tutorials, I would like to let Kamailio handle registrations,
calls between users and other basic functionalities. Asterisk will only
handle advanced features like voicemail, advanced dialplan configuration,.
My problem is that It needs to be multidomain.

My network will be as follow with 2 network interfaces for Kamailio and a
private lan between Kamailio and Asterisk :

Public interface ip 1.2.3.4 > [Kamailio] <- Private interface
192.168.100.10 <> 192.168.100.11 [Asterisk]

When I send a call to asterisk, the domain is sent in the from field of the
INVITE and I can do what I need on the Asterisk dialplan (I can get the SIP
domain using the ${SIPDOMAIN} variable). My problem is when I need to send a
call back to Kamailio for example to reach another user of the domain.

I'm using Asterisk 14 with PJSIP with the following config :

[kamailio]
type=endpoint
transport=transport-udp
context=from-kamailio
disallow=all
allow=ulaw
aors=kamailio

[kamailio]
type=aor
contact=sip:http://192.168.100.10:5060

[kamailio]
type=identify
endpoint=kamailio
match=192.168.100.10

If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio
/mailto:sip%3a...@testdomain.com;, Asterisk contact directly 
http://testcomain.com without
going through the local IP of my Kamailio.

I can't send a domain to Kamailio in the INVITE request.

Does anyone can help me on this or maybe simply tell me that I'm not going
to the good direction? :)

Thank you,

Cyrille


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Re: [SR-Users] Kamailio / Asterisk and multidomain

2017-08-22 Thread Daniel Tryba 
On Tue, Aug 22, 2017 at 02:50:13PM +0200, Cyrille Demaret wrote:
> If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio
> /sip:2...@testdomain.com", Asterisk contact directly testcomain.com without
> going through the local IP of my Kamailio.
> 
> I can't send a domain to Kamailio in the INVITE request.
> 
> Does anyone can help me on this or maybe simply tell me that I'm not going
> to the good direction? :)

Take a look at outbound_proxy for pjsip
https://wiki.asterisk.org/wiki/display/AST/PJSIP+with+Proxies

With chan_sip you could do this dynamically from the dialplan:
; A new feature for 1.8 allows one to specify a host or IP address to use
; when routing the call. This is typically used in tandem with func_srv if
; multiple methods of reaching the same domain exist. The host or IP address
; is specified after the third slash in the dialstring. Examples:
;
; SIP/devicename/extension/IPorHost
; SIP/username@domain//IPorHost


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Re: [SR-Users] Kamailio / Asterisk and multidomain

2017-08-22 Thread Vladyslav Zakhozhai 
Hi,

I think that this issue is more specific for asterisk than kamailio.

Maybe configuration of outbound proxy may help here but I am not sure.

On Tue, Aug 22, 2017 at 3:50 PM, Cyrille Demaret  wrote:

> Hi,
>
> I'm new to Kamailio and I'm trying to use it as a front SIP proxy to one or
> more asterisk. Unlike to the "Kamailio 4.0.x and Asterisk 11.3.0 Realtime
> Integration" tutorials, I would like to let Kamailio handle registrations,
> calls between users and other basic functionalities. Asterisk will only
> handle advanced features like voicemail, advanced dialplan configuration,.
> My problem is that It needs to be multidomain.
>
> My network will be as follow with 2 network interfaces for Kamailio and a
> private lan between Kamailio and Asterisk :
>
> Public interface ip 1.2.3.4 > [Kamailio] <- Private interface
> 192.168.100.10 <> 192.168.100.11 [Asterisk]
>
> When I send a call to asterisk, the domain is sent in the from field of the
> INVITE and I can do what I need on the Asterisk dialplan (I can get the SIP
> domain using the ${SIPDOMAIN} variable). My problem is when I need to send
> a
> call back to Kamailio for example to reach another user of the domain.
>
> I'm using Asterisk 14 with PJSIP with the following config :
>
> [kamailio]
> type=endpoint
> transport=transport-udp
> context=from-kamailio
> disallow=all
> allow=ulaw
> aors=kamailio
>
> [kamailio]
> type=aor
> contact=sip:192.168.100.10:5060
>
> [kamailio]
> type=identify
> endpoint=kamailio
> match=192.168.100.10
>
> If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio
> /sip:2...@testdomain.com", Asterisk contact directly testcomain.com without
> going through the local IP of my Kamailio.
>
> I can't send a domain to Kamailio in the INVITE request.
>
> Does anyone can help me on this or maybe simply tell me that I'm not going
> to the good direction? :)
>
> Thank you,
>
> Cyrille
>
>
>
>
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>



-- 
С уважением,
Владислав Захожай
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[SR-Users] Kamailio / Asterisk and multidomain

2017-08-22 Thread Cyrille Demaret 
Hi,

I'm new to Kamailio and I'm trying to use it as a front SIP proxy to one or
more asterisk. Unlike to the "Kamailio 4.0.x and Asterisk 11.3.0 Realtime
Integration" tutorials, I would like to let Kamailio handle registrations,
calls between users and other basic functionalities. Asterisk will only
handle advanced features like voicemail, advanced dialplan configuration,.
My problem is that It needs to be multidomain.

My network will be as follow with 2 network interfaces for Kamailio and a
private lan between Kamailio and Asterisk : 

Public interface ip 1.2.3.4 > [Kamailio] <- Private interface
192.168.100.10 <> 192.168.100.11 [Asterisk]

When I send a call to asterisk, the domain is sent in the from field of the
INVITE and I can do what I need on the Asterisk dialplan (I can get the SIP
domain using the ${SIPDOMAIN} variable). My problem is when I need to send a
call back to Kamailio for example to reach another user of the domain.

I'm using Asterisk 14 with PJSIP with the following config :

[kamailio]
type=endpoint
transport=transport-udp
context=from-kamailio
disallow=all
allow=ulaw
aors=kamailio

[kamailio]
type=aor
contact=sip:192.168.100.10:5060

[kamailio]
type=identify
endpoint=kamailio
match=192.168.100.10

If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio
/sip:2...@testdomain.com", Asterisk contact directly testcomain.com without
going through the local IP of my Kamailio.

I can't send a domain to Kamailio in the INVITE request.

Does anyone can help me on this or maybe simply tell me that I'm not going
to the good direction? :)

Thank you,

Cyrille




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