Thank you very much for your reply Barry. Can you explain us yours ip 10.5.75. and 10.5.76. ? Seems your asterisks have two interfaces, one private and one public?
In my infrastructure I have: SIP client -> Kamailio ---> Asterisk 01 ---> Asterisk 02 ... ---> Asterisk N
Only Kamailio has a public IP address.
In the REGFWD route you send all the registrations to ALL the Asterisks? If so, why?
I'm trying to achieve something like:
SIP clientA ---> REG ---> Kamailio Is Asterisk01 alive? Yes --> REG ---> Asterisk 01 NO --> Choose next one ---> Asterisk 02 ... ---> Asterisk N
---> Asterisk 01 SIP clientB ---> REG ---> Kamailio Is Asterisl02 alive? Yes --> REG ---> Asterisk 02 NO --> Choose next one ---> Asterisk 03 ... ---> Asterisk N and so on, with a round robin algorithm.
When Sip ClientA want to call Sip ClientB, it asks Kamailio, that know that SIP Client B is registered on Asterisk 02 and so send the invite to it.
I'm going to work over this in the next few days and let you know the result, if positive I'll write a new complete guide on this topic with your valuable suggestions.
Any suggestion is welcome
BR, Anty
------------------------------------------------------------------------------------------------------
I have read and applied the excellent guide found on: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
Now I have added to Kamailio the HA/Load Balancer support, with the "dispatcher" module. This way I have 1 Kamailio and 2 Asterisk machines.
How can I change the routing logic of the sections route[REGFWD] | route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in the guide above the asterisk binded ip address is cabled in the kamailio config code)
I have done that. Relevant route entries below.
# Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; }
# for each Asterisk server in setid=2 (External) we send a registration on behalf of the user. sql_xquery("ca", "SELECT SUBSTRING_INDEX(destination,':',-1) AS port, SUBSTRING_INDEX(SUBSTRING(destination,5),':',1) AS address FROM dispatcher WHERE setid = 2", "ra"); $var(i) = 0; while($xavp(ra[$var(i)]) != $null) { $var(rip) = $xavp(ra[$var(i)]=>address); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $xavp(ra[$var(i)]=>port); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $var(i) = $var(i) + 1; } sql_result_free("ra");
}
# Test if coming from Asterisk. We check the dispatcher "ds_is_from_list()" function to see if this is one of our Asterisk IPs route[FROMASTERISK] { if(ds_is_from_list()) { return 1; } else { return -1; } }
# Send to Asterisk route[TOASTERISK] {
# If call comes in to the .75 iface, we need to send it to the .75 iface of Asterisk as well. # otherwise we send to the .76 iface. We do this by calling different dispatcher sets. This is # because Asterisk needs to use NAT on the .76. (public) interface but not on the .75. if($td=~"10.5.75.") { $var(setid) = 4; xlog("SCRIPT: Call to 10.5.75. ip - using set $var(setid) \n"); } else { $var(setid) = 2; xlog("SCRIPT: Call from $fn to 10.5.76. ip - using set $var(setid) \n"); } # round robin dispatching on set determined above if(!ds_select_dst($var(setid), "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; }
Hope this helps.
-Barry Flanagan
On 30 April 2013 17:23, Aldo Antignano aldo@antignano.it wrote:
Thank you very much for your reply Barry. Can you explain us yours ip 10.5.75. and 10.5.76. ? Seems your asterisks have two interfaces, one private and one public?
Sorry, yeah. you should probably ignore that part :-) The same dispatcher deals with client connections from the "public" network as well as inbound calls from a pstn gateway which is on a "private" network. Asterisk is behind NAT for the public clients (Openstack cloud) so by sending the request on the .76 network Asterisk uses its externhost setting as the IP address for the RTP. Unless you are running this on a virtualized environment it is probably not relevant.
In my infrastructure I have: SIP client -> Kamailio ---> Asterisk 01 ---> Asterisk 02 ... ---> Asterisk N
Only Kamailio has a public IP address.
In the REGFWD route you send all the registrations to ALL the Asterisks? If so, why?
By sending the registration to all the available Asterisk servers then they all know where each SIP client can be located. If a call comes in from the PSTN or from a SIP client Kamailio can just be sent it to any of the * servers and it will find the registered client. This will work even with multiple load balancers/dispatchers using DNS SRV.
In this scenario Kamailio is mainly acting as a load balancer, topology hider and security/anti-flood front-end to create a scalable and redundant Asterisk PBX system. Really depends what you are after. For this particular implementation I was not overly concerned with having too many registrations for it to be a problem for Asterisk, so sending them to all servers was not an issue
Regards,
-Barry Flanagan
I'm trying to achieve something like:
SIP clientA ---> REG ---> Kamailio Is Asterisk01 alive? Yes --> REG ---> Asterisk 01
NO --> Choose next one ---> Asterisk 02 ... ---> Asterisk N ---> Asterisk 01SIP clientB ---> REG ---> Kamailio Is Asterisl02 alive? Yes --> REG ---> Asterisk 02
NO --> Choose next one ---> Asterisk 03 ... ---> Asterisk Nand so on, with a round robin algorithm.
When Sip ClientA want to call Sip ClientB, it asks Kamailio, that know that SIP Client B is registered on Asterisk 02 and so send the invite to it.
I'm going to work over this in the next few days and let you know the result, if positive I'll write a new complete guide on this topic with your valuable suggestions.
Any suggestion is welcome
BR, Anty
I have read and applied the excellent guide found on:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
Now I have added to Kamailio the HA/Load Balancer support, with the "dispatcher" module. This way I have 1 Kamailio and 2 Asterisk machines.
How can I change the routing logic of the sections route[REGFWD] | route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in
the
guide above the asterisk binded ip address is cabled in the kamailio
config
code)
I have done that. Relevant route entries below.
# Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; }
# for each Asterisk server in setid=2 (External) we send a registration on behalf of the user. sql_xquery("ca", "SELECT SUBSTRING_INDEX(destination,':',-1) AS port, SUBSTRING_INDEX(SUBSTRING(destination,5),':',1) AS address FROM dispatcher WHERE setid = 2", "ra"); $var(i) = 0; while($xavp(ra[$var(i)]) != $null) { $var(rip) = $xavp(ra[$var(i)]=>address); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $xavp(ra[$var(i)]=>port); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@"
- $sel(cfg_get.kamailio.bindip)
- ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";
if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $var(i) = $var(i) + 1; } sql_result_free("ra");
}
# Test if coming from Asterisk. We check the dispatcher "ds_is_from_list()" function to see if this is one of our Asterisk IPs route[FROMASTERISK] { if(ds_is_from_list()) { return 1; } else { return -1; } }
# Send to Asterisk route[TOASTERISK] {
# If call comes in to the .75 iface, we need to send it to the .75 iface of Asterisk as well. # otherwise we send to the .76 iface. We do this by calling different dispatcher sets. This is # because Asterisk needs to use NAT on the .76. (public) interface but not on the .75. if($td=~"10.5.75.") { $var(setid) = 4; xlog("SCRIPT: Call to 10.5.75. ip - using set $var(setid) \n"); } else { $var(setid) = 2; xlog("SCRIPT: Call from $fn to 10.5.76. ip - using set $var(setid) \n"); } # round robin dispatching on set determined above if(!ds_select_dst($var(setid), "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; }
Hope this helps.
-Barry Flanagan
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Aldo Antignano -
Barry Flanagan