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:200@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
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 cyrille@omail.be 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:200@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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
I feel stupid but indeed setting up Kamelio as proxy in Asterisk did the trick.
Regards,
Cyrille
De : sr-users [mailto:sr-users-bounces@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 mailto:cyrille@omail.be 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%3A200@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
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:200@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
-
Cyrille Demaret -
Daniel Tryba -
Vladyslav Zakhozhai