20 Aug
2019
20 Aug
'19
7:22 p.m.
All,
I would like to use Kamailio as a frontend/proxy for an asterisk
machine/cluster. I'm very well versed in Asterisk, but VERY noob to
Kamailio.
I've watched so many Kamailio presentations and feel like I've scoured
the web for any article that could help me. There is alot out there, but
I'm not finding my exact situation.
I believe I understand the dispatcher stuff that I'll need down the
road, but I'm having issues understanding the setup of my SIP trunk and
Kamailio.
I used UAC to setup the trunk connection in Kamailio and it sort of
worked. I'm not sure what I do for Asterisk. I'd suspect that in this
situation Kamailio would handle the registration and trunk connection
for me.
What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need
to still register the trunk from each Asterisk box "thru" the Kamailio
proxy, etc?
Also, I'm merely accepting outside calls and then validating the caller
and bridging them back out to the PSTN, so I don't have any local SIP
clients, etc., so no need to register the sip devices, etc.
Thanks!
Travis
20 Aug
20 Aug
7:54 p.m.
Hello,
You can just set the outbound proxy on asterisk to use kamailio. Thats
should work properly. Kamailio will simply act as a proxy.
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
On Tue, Aug 20, 2019 at 3:23 PM Travis Ryan <travis(a)travisryan.com> wrote:
All,
I would like to use Kamailio as a frontend/proxy for an asterisk
machine/cluster. I'm very well versed in Asterisk, but VERY noob to
Kamailio.
I've watched so many Kamailio presentations and feel like I've scoured
the web for any article that could help me. There is alot out there, but
I'm not finding my exact situation.
I believe I understand the dispatcher stuff that I'll need down the
road, but I'm having issues understanding the setup of my SIP trunk and
Kamailio.
I used UAC to setup the trunk connection in Kamailio and it sort of
worked. I'm not sure what I do for Asterisk. I'd suspect that in this
situation Kamailio would handle the registration and trunk connection
for me.
What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need
to still register the trunk from each Asterisk box "thru" the Kamailio
proxy, etc?
Also, I'm merely accepting outside calls and then validating the caller
and bridging them back out to the PSTN, so I don't have any local SIP
clients, etc., so no need to register the sip devices, etc.
Thanks!
Travis
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
8:01 p.m.
On Tue, Aug 20, 2019 at 10:22:26AM -0400, Travis Ryan wrote:
What role is Kamailio to my Asterisk? Just an Outbound
proxy? Do I need to
still register the trunk from each Asterisk box "thru" the Kamailio proxy,
etc?
Also, I'm merely accepting outside calls and then validating the caller and
bridging them back out to the PSTN, so I don't have any local SIP clients,
etc., so no need to register the sip devices, etc.
The real question is what do you need kamailio to do? You answer this
with as a simple proxy.
A possible solution for you is to use kamilio with the dispatcher module. One
id (1) for the PSTN side, one id (2) for the Asterisk side. If a call comes in
from 1, route it to 2 and v.v.
This makes the kamailio machine the "endpoint" for both PSTN and
Asterisk side.
With the "default" config that comes with kamailio all you need to do is
strip out anything from the accounting bit in request_route (line 508)
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
and insert something like:
if(ds_is_from_list("1",3))
{
$avp(dispatcherid)="2";
}
else if(ds_is_from_list("2",3))
{
$avp(dispatcherid)="1";
}
else
{
send_reply("403", "Go away");
exit;
}
route(DISPATCHER);
route(RELAY);
With route DISPATCHER being:
route[DISPATCHER]
{
if(!ds_select_dst($avp(dispatcherid), "4"))
{
send_reply("501", "No dispatcher available");
exit;
}
t_on_failure("RTF_DISPATCH");
return;
}
See https://kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher…
for more info on integrating the dispatcher module.
More advanced subjects to look at are:
-do you need an rtp proxy?
-do you need topology hiding?
-is NAT involved?
But leave them until you have a clue about how to use kamailio as a sip proxy in a
simple test environment (e.g. between 2 asterisk servers).
8:10 p.m.
Thanks,
I want to eventually get to a setup like the one here:
https://github.com/CyCoreSystems/asterisk-k8s-demo
But since I'll need Kamailio to handle a high load of incoming calls, I
think I need it to direct traffic, etc for any number of Asterisk
servers behind it.
In this setup it indeed has RTPProxy, etc. I just want to understand how
to use it rather than just drop it in, etc. Also the demo doesn't have
any config for an outside SIP trunk, etc.
Maybe this helps?
Thanks,
Travis
On 8/20/19 11:01 AM, Daniel Tryba wrote:
On Tue, Aug 20, 2019 at 10:22:26AM -0400, Travis Ryan
wrote:
What role is Kamailio to my Asterisk? Just an
Outbound proxy? Do I need to
still register the trunk from each Asterisk box "thru" the Kamailio proxy,
etc?
Also, I'm merely accepting outside calls and then validating the caller and
bridging them back out to the PSTN, so I don't have any local SIP clients,
etc., so no need to register the sip devices, etc.
The real question is what do
you need kamailio to do? You answer this
with as a simple proxy.
A possible solution for you is to use kamilio with the dispatcher module. One
id (1) for the PSTN side, one id (2) for the Asterisk side. If a call comes in
from 1, route it to 2 and v.v.
This makes the kamailio machine the "endpoint" for both PSTN and
Asterisk side.
With the "default" config that comes with kamailio all you need to do is
strip out anything from the accounting bit in request_route (line 508)
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
and insert something like:
if(ds_is_from_list("1",3))
{
$avp(dispatcherid)="2";
}
else if(ds_is_from_list("2",3))
{
$avp(dispatcherid)="1";
}
else
{
send_reply("403", "Go away");
exit;
}
route(DISPATCHER);
route(RELAY);
With route DISPATCHER being:
route[DISPATCHER]
{
if(!ds_select_dst($avp(dispatcherid), "4"))
{
send_reply("501", "No dispatcher available");
exit;
}
t_on_failure("RTF_DISPATCH");
return;
}
See
https://kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher…
for more info on integrating the dispatcher module.
More advanced subjects to look at are:
-do you need an rtp proxy?
-do you need topology hiding?
-is NAT involved?
But leave them until you have a clue about how to use kamailio as a sip proxy in a
simple test environment (e.g. between 2 asterisk servers).
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
21 Aug
21 Aug
7:08 p.m.
Hi Travis,
two projects that enables siptrunking and routing:
https://github.com/voiceboys/sbcOS (ip-auth based or registrar on your
side)
https://dsiprouter.readthedocs.io/en/latest/
Also intressting to see how they solved this problems.
If you could describe your siptrunk a bit more,
then would here many people that can point you in the right direction how
to solve that with or without kamailio.
Cheers
Karsten
Am Di., 20. Aug. 2019 um 17:11 Uhr schrieb Travis Ryan <
travis(a)travisryan.com>gt;:
Thanks,
I want to eventually get to a setup like the one here:
https://github.com/CyCoreSystems/asterisk-k8s-demo
But since I'll need Kamailio to handle a high load of incoming calls, I
think I need it to direct traffic, etc for any number of Asterisk
servers behind it.
In this setup it indeed has RTPProxy, etc. I just want to understand how
to use it rather than just drop it in, etc. Also the demo doesn't have
any config for an outside SIP trunk, etc.
Maybe this helps?
Thanks,
Travis
On 8/20/19 11:01 AM, Daniel Tryba wrote:
module. One
--
Mit freundlichen Grüßen
*Karsten Horsmann*
On Tue, Aug 20, 2019 at 10:22:26AM -0400, Travis
Ryan wrote:
> What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need
to
> still register the trunk from each Asterisk
box "thru" the Kamailio
proxy,
> etc?
>
> Also, I'm merely accepting outside calls and then validating the caller
and
> bridging them back out to the PSTN, so I
don't have any local SIP
clients,
etc., so
no need to register the sip devices, etc.
The real question is what do you need
kamailio to do? You answer this
with as a simple proxy.
A possible solution for you is to use kamilio with the dispatcher id (1) for the PSTN side, one id (2) for the
Asterisk side. If a call
comes in
from 1, route it to 2 and v.v.
This makes the kamailio machine the "endpoint" for both PSTN and
Asterisk side.
With the "default" config that comes with kamailio all you need to do is
strip out anything from the accounting bit in request_route (line 508)
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
and insert something like:
if(ds_is_from_list("1",3))
{
$avp(dispatcherid)="2";
}
else if(ds_is_from_list("2",3))
{
$avp(dispatcherid)="1";
}
else
{
send_reply("403", "Go away");
exit;
}
route(DISPATCHER);
route(RELAY);
With route DISPATCHER being:
route[DISPATCHER]
{
if(!ds_select_dst($avp(dispatcherid), "4"))
{
send_reply("501", "No dispatcher available");
exit;
}
t_on_failure("RTF_DISPATCH");
return;
}
See
https://kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher…
for more info on integrating the dispatcher
module.
More advanced subjects to look at are:
-do you need an rtp proxy?
-do you need topology hiding?
-is NAT involved?
But leave them until you have a clue about how to use kamailio as a sip
proxy in
a
simple test environment (e.g. between 2 asterisk
servers).
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
22 Aug
22 Aug
5:56 p.m.
I'm using sip.digiumcloud.net but eventually will be using wholesale at
bandwidth.com. I'll take a look at your links. Thanks!
On 8/21/19 10:08 AM, Karsten Horsmann wrote:
Hi Travis,
two projects that enables siptrunking and routing:
https://github.com/voiceboys/sbcOS (ip-auth based or registrar on your
side)
https://dsiprouter.readthedocs.io/en/latest/
Also intressting to see how they solved this problems.
If you could describe your siptrunk a bit more,
then would here many people that can point you in the right direction
how to solve that with or without kamailio.
Cheers
Karsten
Am Di., 20. Aug. 2019 um 17:11 Uhr schrieb Travis Ryan
<travis(a)travisryan.com <mailto:travis@travisryan.com>>:
Thanks,
I want to eventually get to a setup like the one here:
https://github.com/CyCoreSystems/asterisk-k8s-demo
But since I'll need Kamailio to handle a high load of incoming
calls, I
think I need it to direct traffic, etc for any number of Asterisk
servers behind it.
In this setup it indeed has RTPProxy, etc. I just want to
understand how
to use it rather than just drop it in, etc. Also the demo doesn't
have
any config for an outside SIP trunk, etc.
Maybe this helps?
Thanks,
Travis
On 8/20/19 11:01 AM, Daniel Tryba wrote:
this
On Tue, Aug 20, 2019 at 10:22:26AM -0400, Travis
Ryan wrote:
> What role is Kamailio to my Asterisk? Just an Outbound proxy?
Do I need
to
> still register the trunk from each Asterisk
box "thru" the
Kamailio proxy,
> etc?
>
> Also, I'm merely accepting outside calls and then validating
the
caller and
> bridging them back out to the PSTN, so I
don't have any local
SIP clients,
etc., so
no need to register the sip devices, etc.
The real question is what do you need
kamailio to do? You answer with as a simple proxy.
A possible solution for you is to use kamilio with the
dispatcher module. One
id (1) for the PSTN side, one id (2) for the
Asterisk side. If a
call comes in
from 1, route it to 2 and v.v.
This makes the kamailio machine the "endpoint" for both PSTN and
Asterisk side.
With the "default" config that comes with kamailio all you need
to
do is
strip out anything from the accounting bit in
request_route
(line 508)
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
and insert something like:
if(ds_is_from_list("1",3))
{
$avp(dispatcherid)="2";
}
else if(ds_is_from_list("2",3))
{
$avp(dispatcherid)="1";
}
else
{
send_reply("403", "Go away");
exit;
}
route(DISPATCHER);
route(RELAY);
With route DISPATCHER being:
route[DISPATCHER]
{
if(!ds_select_dst($avp(dispatcherid), "4"))
{
send_reply("501", "No dispatcher available");
exit;
}
t_on_failure("RTF_DISPATCH");
return;
}
See
https://kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher…
for more info on integrating the dispatcher
module.
More advanced subjects to look at are:
-do you need an rtp proxy?
-do you need topology hiding?
-is NAT involved?
But leave them until you have a clue about how to use kamailio
as a sip proxy
in a
simple test environment (e.g. between 2 asterisk
servers).
_______________________________________________
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
_______________________________________________
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
--
Mit freundlichen Grüßen
*Karsten Horsmann*
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
1720
days inactive
1722
days old
5 comments
4 participants
participants (4)
-
Daniel Tryba -
David Villasmil -
Karsten Horsmann -
Travis Ryan