12 Jun
2017
12 Jun
'17
10:21 p.m.
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.
13 Jun
13 Jun
10:01 a.m.
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.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
14 Jun
14 Jun
7:17 p.m.
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.
_______________________________________________
Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
7:45 p.m.
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 <mailto:miconda@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:
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
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.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
--
Daniel-Constantin Mierla
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio Advanced Training - www.asipto.com <http://www.asipto.com>
Kamailio World Conference - www.kamailioworld.com
<http://www.kamailioworld.com>
_______________________________________________ Kamailio (SER) -
Users Mailing List sr-users(a)lists.kamailio.org
<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
15 Jun
15 Jun
12:28 a.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.
_______________________________________________
Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
_______________________________________________ Kamailio (SER) - Users
Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi
-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
11:24 a.m.
On Wed, Jun 14, 2017 at 05:28:35PM -0400, Jean Cérien wrote:
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.
and the invite arriving
INVITE sip:1100@192.168.2.228 SIP/2.0
Any help will be greatly appreciated
Either accept all INVITEs from the IPs you are registering, use
something like db_aliases to map 1100 to your known username or change
the setup of the PBX you are registering to to not use the dialed nummer
in the R-URI but the username (it is an asterisk so it is the difference
between Dial(SIP/${EXTEN}@username) and Dial(SIP/username))
16 Jun
16 Jun
6:55 p.m.
Thanks.
I dont have control over the host I need to register to, so I cant
unfortunately retrieve the luuid.
Is there a way to lookup in the uacreg table with the IP ? I could only
find uac_reg_lookup, but it required the luuid. I've worked around that
adding an entry into address, but I am not really satisfied with this
method,
Regards,
J
On Thu, Jun 15, 2017 at 4:24 AM, Daniel Tryba <d.tryba(a)pocos.nl> wrote:
On Wed, Jun 14, 2017 at 05:28:35PM -0400, Jean Cérien
wrote:
I have the registration working, but when my
kamailio receives an
INVITE, I
am a bit lost regarding what to do. in my
route(AUTH), I am trying to
check
uac_reg_status, but no success.
and the invite arriving
INVITE sip:1100@192.168.2.228 SIP/2.0
Any help will be greatly appreciated
Either accept all INVITEs from the IPs you are registering, use
something like db_aliases to map 1100 to your known username or change
the setup of the PBX you are registering to to not use the dialed nummer
in the R-URI but the username (it is an asterisk so it is the difference
between Dial(SIP/${EXTEN}@username) and Dial(SIP/username))
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
19 Jun
19 Jun
2:31 p.m.
On Fri, Jun 16, 2017 at 11:55:06AM -0400, Jean Cérien wrote:
Is there a way to lookup in the uacreg table with the
IP ? I could only
find uac_reg_lookup, but it required the luuid. I've worked around that
adding an entry into address, but I am not really satisfied with this
method,
You could use slqops to query the database manually.
2716
days inactive
2723
days old
7 comments
3 participants
participants (3)
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Daniel Tryba -
Daniel-Constantin Mierla -
Jean Cérien