[SR-Users] Kamailio Infront of Asterisk with remote PBX

[Kenny Watson]

Hi,

I have a few asterisk servers providing some basic SIP trunking and routing.

We have remote PBXs trunked onto asterisk which calls come into asterisk and are routing down to extensions on the remote PBX via prefix routing.

I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer which Invites come into and are routed out to either SIP phones which are registered or to the remote PBX.

I'm looking for some advice as to which modules would be best to use to achieve this as the remote PBXs will be dynamically registered rather than fixed gateways.

Please let me know what further information would be helpful.

Thanks
Kenny Watson

[Fred Posner]

Hi Kenny,

This depends on the carriers and scenarios that you may use. I know
"depends" is a horrible answer, but one of the great aspects of Kamailio
is the flexibility of the modules.

Some deployments may have a group of Asterisk servers all configured
similarly for handling calls. With this type of scenario, you would
benefit from using the dispatcher module.

Many people like to use Kamailio on the public side of their network and
keep their asterisk servers on the private. This would be an example of
when to use rtpproxy (in bridge mode).

Some carriers hate seeing the chain of systems on your network (ie the
asterisk boxes). Sometimes the use of TOPOH helps to integrate with the
carriers who have chosen their own "interpretations" of RFC for "security."

And there's more...

The bottom line, is that the devil is in the details.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
[phone] (direct)
[phone] (fax)

Post by Kenny Watson
Hi,
I have a few asterisk servers providing some basic SIP trunking and routing.
We have remote PBXs trunked onto asterisk which calls come into asterisk
and are routing down to extensions on the remote PBX via prefix routing.
I’m looking to have a central Kamailio Registrar/Proxy/Loadbalancer
which Invites come into and are routed out to either SIP phones which
are registered or to the remote PBX.
I’m looking for some advice as to which modules would be best to use to
achieve this as the remote PBXs will be dynamically registered rather
than fixed gateways.
Please let me know what further information would be helpful.
Thanks
Kenny Watson

[Kenny Watson]

This post might be inappropriate. Click to display it.

[Fred Posner]

If you want to call a user on Kamailio from Asterisk...

example...

exten => s,1,Verbose(4,calling user on kamailio)
same => n,Dial(SIP/***@KAMAILIO,time,options)
same => n,--after dial logic --


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
[phone] (direct)
[phone] (fax)

Post by Kenny Watson
Hi Fred,
Thanks for the quick response. I already do use some Kamailio
features on our internal network for load balancing.
The use case that I'm interested in is to effectively replace an
asterisk server that I use for SIP trunking to remote phone systems
with a Kamailio registrar/proxy and a bank of asterisk servers
placing calls direct to extensions on the remote PBX.
I currently have this running on asterisk which I route to the
different remote PBX extensions using prefix based routing down to
the destination peer on asterisk which is essentially what I need to
replicate on Kamailio.
i.e.
remotepbx1 maybe defined as either by IP address or via a "normal"
registered sip peer with a username/password combo.
I understand that I can dial a registered device directly but its how
to call a remote extension on a registered device via Kamailio.
Thanks Kenny Watson
Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX
Hi Kenny,
This depends on the carriers and scenarios that you may use. I know
"depends" is a horrible answer, but one of the great aspects of
Kamailio is the flexibility of the modules.
Some deployments may have a group of Asterisk servers all configured
similarly for handling calls. With this type of scenario, you would
benefit from using the dispatcher module.
Many people like to use Kamailio on the public side of their network
and keep their asterisk servers on the private. This would be an
example of when to use rtpproxy (in bridge mode).
Some carriers hate seeing the chain of systems on your network (ie
the asterisk boxes). Sometimes the use of TOPOH helps to integrate
with the carriers who have chosen their own "interpretations" of RFC
for "security."
And there's more...
The bottom line, is that the devil is in the details.
Fred Posner The Palner Group, Inc. http://www.palner.com (web)
[phone] (direct) [phone] (fax)

Post by Kenny Watson
Hi,
I have a few asterisk servers providing some basic SIP trunking and routing.
We have remote PBXs trunked onto asterisk which calls come into
asterisk and are routing down to extensions on the remote PBX via
prefix routing.
I'm looking to have a central Kamailio
Registrar/Proxy/Loadbalancer which Invites come into and are routed
out to either SIP phones which are registered or to the remote
PBX.
I'm looking for some advice as to which modules would be best to
use to achieve this as the remote PBXs will be dynamically
registered rather than fixed gateways.
Please let me know what further information would be helpful.
Thanks
Kenny Watson

[Kenny Watson]

Hi Fred,

Its more that the "user" on Kamailio is actually a PBX with extensions on it.

On asterisk I'd usually do Dial(SIP/peername/extension) but I obviously cant do this as Kamailio is the peer that the call is being routed to initially.

What I need to figure out is how to on kamailo maybe using a dial prefix specify that the call is going to a remote extension on a "user" and rewrite the to header to be ***@useripaddress rather than ***@useripaddress.

Does this make sense?


Thanks
Kenny Watson

-----Original Message-----
From: sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org [mailto:sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org] On Behalf Of Fred Posner
Sent: 23 October 2014 16:34
To: sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org
Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX

If you want to call a user on Kamailio from Asterisk...

example...

exten => s,1,Verbose(4,calling user on kamailio)
same => n,Dial(SIP/***@KAMAILIO,time,options)
same => n,--after dial logic --


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
[phone] (direct)
[phone] (fax)

Post by Kenny Watson
Hi Fred,
Thanks for the quick response. I already do use some Kamailio
features on our internal network for load balancing.
The use case that I'm interested in is to effectively replace an
asterisk server that I use for SIP trunking to remote phone systems
with a Kamailio registrar/proxy and a bank of asterisk servers placing
calls direct to extensions on the remote PBX.
I currently have this running on asterisk which I route to the
different remote PBX extensions using prefix based routing down to the
destination peer on asterisk which is essentially what I need to
replicate on Kamailio.
i.e.
remotepbx1 maybe defined as either by IP address or via a "normal"
registered sip peer with a username/password combo.
I understand that I can dial a registered device directly but its how
to call a remote extension on a registered device via Kamailio.
Thanks Kenny Watson
Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX
Hi Kenny,
This depends on the carriers and scenarios that you may use. I know
"depends" is a horrible answer, but one of the great aspects of
Kamailio is the flexibility of the modules.
Some deployments may have a group of Asterisk servers all configured
similarly for handling calls. With this type of scenario, you would
benefit from using the dispatcher module.
Many people like to use Kamailio on the public side of their network
and keep their asterisk servers on the private. This would be an
example of when to use rtpproxy (in bridge mode).
Some carriers hate seeing the chain of systems on your network (ie the
asterisk boxes). Sometimes the use of TOPOH helps to integrate with
the carriers who have chosen their own "interpretations" of RFC for
"security."
And there's more...
The bottom line, is that the devil is in the details.
Fred Posner The Palner Group, Inc. http://www.palner.com (web)
[phone] (direct) [phone] (fax)

Post by Kenny Watson
Hi,
I have a few asterisk servers providing some basic SIP trunking and routing.
We have remote PBXs trunked onto asterisk which calls come into
asterisk and are routing down to extensions on the remote PBX via
prefix routing.
I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer
which Invites come into and are routed out to either SIP phones which
are registered or to the remote PBX.
I'm looking for some advice as to which modules would be best to use
to achieve this as the remote PBXs will be dynamically registered
rather than fixed gateways.
Please let me know what further information would be helpful.
Thanks
Kenny Watson

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[Ovidiu Sas]

You can implement whatever routing logic you want in kamailio.
Also, you can use different sets of modules to implement same type of
routing logic.

The To header is irrelevant in SIP, no need for re-write.
When a call is received from an endpoint, based on it's IP address you
can choose what to do with the call (for example: find the right user
for the given extension).

You need to figure out all the requirements for your new setup and
then start implementing it.

Since you already have kamailio servers in your setup, you are
familiar with how kamailio works.
All you need to do is do some more modules README reading and figure
out which one is a better fit for your needs.


Regards,
Ovidiu Sas

Post by Kenny Watson
Hi Fred,
Its more that the "user" on Kamailio is actually a PBX with extensions on it.
On asterisk I'd usually do Dial(SIP/peername/extension) but I obviously cant do this as Kamailio is the peer that the call is being routed to initially.
Does this make sense?
Thanks
Kenny Watson
-----Original Message-----
Sent: 23 October 2014 16:34
Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX
If you want to call a user on Kamailio from Asterisk...
example...
exten => s,1,Verbose(4,calling user on kamailio)
same => n,--after dial logic --
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
[phone] (direct)
[phone] (fax)

Post by Kenny Watson
Hi Fred,
Thanks for the quick response. I already do use some Kamailio
features on our internal network for load balancing.
The use case that I'm interested in is to effectively replace an
asterisk server that I use for SIP trunking to remote phone systems
with a Kamailio registrar/proxy and a bank of asterisk servers placing
calls direct to extensions on the remote PBX.
I currently have this running on asterisk which I route to the
different remote PBX extensions using prefix based routing down to the
destination peer on asterisk which is essentially what I need to
replicate on Kamailio.
i.e.
remotepbx1 maybe defined as either by IP address or via a "normal"
registered sip peer with a username/password combo.
I understand that I can dial a registered device directly but its how
to call a remote extension on a registered device via Kamailio.
Thanks Kenny Watson
Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX
Hi Kenny,
This depends on the carriers and scenarios that you may use. I know
"depends" is a horrible answer, but one of the great aspects of
Kamailio is the flexibility of the modules.
Some deployments may have a group of Asterisk servers all configured
similarly for handling calls. With this type of scenario, you would
benefit from using the dispatcher module.
Many people like to use Kamailio on the public side of their network
and keep their asterisk servers on the private. This would be an
example of when to use rtpproxy (in bridge mode).
Some carriers hate seeing the chain of systems on your network (ie the
asterisk boxes). Sometimes the use of TOPOH helps to integrate with
the carriers who have chosen their own "interpretations" of RFC for
"security."
And there's more...
The bottom line, is that the devil is in the details.
Fred Posner The Palner Group, Inc. http://www.palner.com (web)
[phone] (direct) [phone] (fax)

Post by Kenny Watson
Hi,
I have a few asterisk servers providing some basic SIP trunking and routing.
We have remote PBXs trunked onto asterisk which calls come into
asterisk and are routing down to extensions on the remote PBX via
prefix routing.
I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer
which Invites come into and are routed out to either SIP phones which
are registered or to the remote PBX.
I'm looking for some advice as to which modules would be best to use
to achieve this as the remote PBXs will be dynamically registered
rather than fixed gateways.
Please let me know what further information would be helpful.
Thanks
Kenny Watson

_______________________________________________
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users