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/USERNAME@KAMAILIO,time,options) same => n,--after dial logic --

Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax)

On 10/23/2014 11:21 AM, Kenny Watson wrote:

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.

2021XXXX routes to XXXX@remotepbx1

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

-----Original Message----- From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Fred Posner Sent: 23 October 2014 16:00 To: sr-users@lists.sip-router.org 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) +1-503-914-0999 (direct) +1-954-472-2896 (fax)

On 10/23/2014 09:12 AM, Kenny Watson wrote:

...

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

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

On Thu, Oct 23, 2014 at 1:16 PM, Kenny Watson KWatson@geniusppt.com wrote:

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 extension@useripaddress rather than user@useripaddress.

Does this make sense?

Thanks Kenny Watson

-----Original Message----- From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Fred Posner Sent: 23 October 2014 16:34 To: sr-users@lists.sip-router.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/USERNAME@KAMAILIO,time,options) same => n,--after dial logic --

Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax)

On 10/23/2014 11:21 AM, Kenny Watson wrote:

...

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.

2021XXXX routes to XXXX@remotepbx1

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

-----Original Message----- From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Fred Posner Sent: 23 October 2014 16:00 To: sr-users@lists.sip-router.org 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) +1-503-914-0999 (direct) +1-954-472-2896 (fax)

On 10/23/2014 09:12 AM, Kenny Watson wrote:

...

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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

---

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 Daniel,

thanks for this, it looks like exactly what I'm looking for. I've been moved onto another project for the next week but will try this as soon as I can.

Thanks Kenny Watson

-----Original Message----- From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel Tryba Sent: 24 October 2014 12:40 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX

On Thursday 23 October 2014 19:16:58 Kenny Watson wrote:

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 extension@useripaddress rather than user@useripaddress.

I think your "problem" can be summarized as:

initial trunk invite: INVITE sip:number@kamailio in kamailio $rU is number, but after a successful location lookup (lookup("location")/alias_db_lookup) $rU is the username from subscribers or dbaliases. Without further processing the invite to asterisk would look like: INVITE sip:username@asterisk

The solution I use is simple: -before lookup I store the initial $rU in an avp(callednumber). -lookup -check in usr_preferences wether the endpoint wants DID or should be addressed with its username (true/false) -if true: $rU=$avp(callednumber) -t_relay()