Re: [SR-Users] [Kamailio-users] Weight based routing
On Tue, Jul 27, 2010 at 1:13 PM, marius zbihlei marius.zbih...@1and1.rowrote: Alex Balashov wrote: Simplest thing to do would be to use 'dispatcher' with round-robin strategy and enter the servers in the table or list file in duplicate, in proportion to the weight allocation they have. On 07/27/2010 01:55 AM, Chandrakant Solanki wrote: Hello Also the carrierroute module has this capability. Cheers Marius Hello, I have setup 3 SIP server. It should be Kamailio, Asterisk or Any SIP Server. I have setup 3 server say Server A, Server B and Server C and each has weight like 50, 30, 20 in percentage. And I taken 10 calls and try to forward call based on weight. Is this possible in kamailio's module.. and if yes then which module will help me? Any Idea.. -- Regards, Chandrakant Solanki ___ 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 Hi Thanks for reply. I have look into carrierroute module, it seems that my requirement is fulfill. But i have multiple carrier in my carrier_name table, say car1, car2, car3, default, and I want them load dynamically on INVITE of any user based on its probability. Above all carrier has 0.1 probability and each point to different asterisk machine. How could it be possible..?? Any Idea..? -- Regards, Chandrakant Solanki ___ 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
Re: [SR-Users] [Kamailio-users] Weight based routing
Chandrakant Solanki wrote:
Hi
Thanks for reply.
I have look into carrierroute module, it seems that my requirement is
fulfill.
But i have multiple carrier in my carrier_name table, say car1, car2,
car3, default, and I want them load dynamically on INVITE of any user
based on its probability. Above all carrier has 0.1 probability and
each point to different asterisk machine.
How could it be possible..??
Any Idea..?
--
Regards,
Chandrakant Solanki
Hello,
I think you can do things much simpler . For example you don't event
need a database, you could use carrierroute in file mode . Just have one
sp-route.conf file containing the entries like in this example:
domain my_domain {
prefix NULL {
max_targets = 10
target asterisk_server1:5060 {
prob = 0.1
hash_index = 1
status = 1
comment = asterisk server 1
}
target asterisk_server2:5060 {
prob = 0.1
hash_index = 2
status = 1
comment = aserisk server 2
}
.
}
}
Then in the cfg file you could specify modparam(carrierroute,
config_file, /path/to/sp-route.conf)
To balance INVITEs, in you cfg you call
if(method == INVITE){
cr_route(default, my_domain, $rU, $rU, call_id);
forward();
}
This will balance your invites based on the Call-id header.(it will be
sent randomly to one of the hosts based on the weight). If you want you
could balance them based on from_user(Call originating from one user
will always go to the same asterisk machine). Note that in file mode you
only have 1 carrier(called default), but you can have several domains.
Hope this helps
Marius
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Re: [SR-Users] Loops using uac_replace_from
Hello,
On Wed, Jul 28, 2010 at 03:07:20PM +0200, Daniel-Constantin Mierla wrote:
On 7/28/10 2:56 PM, Ján ONDREJ (SAL) wrote:
after some failures with uac_auth (uac_auth can't increase CSeq number,
which is required by my provider) my provider maked me an testing access
without autentification, but I still can't make calls with kamailio or
openser. I can make calls with opensips, but I am still not chosen, which
sip software to use in future (still using openser).
This is a part of my configuration:
route[6] {
uac_replace_from(sip:556807...@as.xx.sk);
rewritehostport(122.33.44.200);
route(5);
}
When trying to sniff packets and diff them, I am still getting this from my
kamailio server:
U kamailio_ip:5060 - client_ip:5060
SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL).
From: Ondrej Jansip:xxx...@;tag=59427D6F-81E63720.
To:sip:xx...@x;user=phone;tag=cc70c0fe52d7af0accfb78a94d592620.f69c.
CSeq: 1 INVITE.
Call-ID: 37cadcfb-f93b8e1d-431c...@xxx.
Server: kamailio (3.0.2 (i386/linux)).
Content-Length: 0.
Logs are below. Any idea, what happening and why a default opensips
configuration works and kamailio/openser don't?
what is your route[5]? It doesn't look at all like being default
config for kamailio anyhow. There seem to be two problems in your
config:
- one is that you try to re-forward without appending a new branch
in failure case (upcoming 3.1 has a auto-detection if there is a new
destination so append_branch() won't be needed)
- you have a recursive calling of route blocks, like route(5)
calling route(6) which calls again route (5)
This cyclic route typo was my problem. Thank you for help.
SAL
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