14 Jun
2017
14 Jun
'17
11:28 p.m.
Thanks
I have the registration working, but when my kamailio receives an INVITE, I
am a bit lost regarding what to do. in my route(AUTH), I am trying to check
uac_reg_status, but no success.
my db:
kamailio:/etc/kamailio# kamcmd uac.reg_dump
{
l_uuid: 123
l_username: passgwada
l_domain: 192.168.2.228
r_username: passgwada
r_domain: 192.168.2.230
realm: xivo
auth_username: passgwada
auth_password: 1234
auth_proxy: sip:192.168.2.230
expires: 60
flags: 2
diff_expires: -1497475389
timer_expires: 0
reg_init: 1497475386
reg_delay: 5
}
and the invite arriving
INVITE sip:1100@192.168.2.228 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.230:5060;branch=z9hG4bK17eeae32;rport
Max-Forwards: 70
From: "0512345678" <sip:0512345678@192.168.2.230>;tag=as0541bcd3
To: <sip:1100@192.168.2.228>
Contact: <sip:0512345678@192.168.2.230:5060>
Call-ID: 0bf6e5ac4fe10a0e3478a6dc1c70d40d@192.168.2.230:5060
CSeq: 102 INVITE
User-Agent: XiVO PBX
Date: Wed, 14 Jun 2017 21:09:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300
......
Any help will be greatly appreciated
J.
On Wed, Jun 14, 2017 at 12:45 PM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Sipsak or other tool pointing via Contact header to
Kamailio is still an
option, but you can use directly uac module -- you have to set the details
for remote registrations inside uac_reg database table and configure uac
module, some details also at:
- https://kamailio.org/docs/modules/5.0.x/modules/uac.html#idp38102932
Cheers,
Daniel
On 14.06.17 18:17, Jean Cérien wrote:
Thanks, I went for the permission + dispatcher, and so far it looks good.
One last thing.... I need the kamailio server to explicitly register to a
provider. Googling the archive, I could only find an old reference to use
sipsak. Does that mean that sipsak (or any other sip crafting tool like
sipsak) should send a register request to Kamailio, which in turn will
forward it to the voip provider ?
Here is the message: https://lists.kamailio.org/pipermail/sr-
users/2004-November/041304.html
Rgds
J
On Tue, Jun 13, 2017 at 3:01 AM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Hello,
I typically blend several modules to achieve your routing needs:
- permissions module with address table to detect if calls are coming
from providers or asterisk farm
- based on result use load balancer to send out to asterisk farm
- or use pdt (if each provider has one gateway) or mtree+dispatcher
(mtree to translate from a prefix to a dispatcher set id of the provider
with many gateways) to send out to providers
If least cost routing is needed towards providers, then you can decide to
use one of next modules:
- lcr
- carrierroute
- drouting
Cheers,
Daniel
On 12.06.17 21:21, Jean Cérien wrote:
Hello
I am looking at the following configuration:
1/ voip providers, with different external sip servers, will send calls
to kamailio, that will dispatch them to asterisk servers, with load
balancing & failover. The IP of the asterisk servers are known and fixed.
2/ the asterisk servers will also place calls, going through kamailio,
that will dispatch them to the voip provider sip servers, based on R-URI /
Prefix.
I've achieved #1 with dispatcher module, I've seen carrierroute module
that can achieve #2, but I dont really grasp how to put the two together.
I have the feeling that #1 can also be done with carrierroute, and that I
only need to check if the 'from' domain is one my asterisk, and chose a
different routing table, but I dont see how to do that
is that a valid approch ? where should I look to do that ?
thanks
J.
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