[OpenSIPS-Users] How to implement a SIP Trunk in between twoSIP servers.

Wed Jan 12 04:18:23 CET 2011

Hi Steven,

To configure the trunk in CUCM, go to Device > Trunk, add a new "SIP
trunk".

The configuration fields are pretty straight forward. Important ones are

*         Destination Address, i.e. opensips IP

*         Port, if not 5060

*         CSS for inbound and outbound calls. (this decide what number
you can send calls to and receive calls from opensips)

*         Any number transformation if you have

This is the basic. If you have questions about particular fields, please
mail in details.

Regards,

Leon

From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of steven chew
Sent: Tuesday, 11 January 2011 11:50 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] How to implement a SIP Trunk in between
twoSIP servers.

Hi Bogdan,

Thanks for your reply.

Your script is very useful for calling between two opensips servers
which I have tested.

However, I don't know how to configure on CUCM 7.0 which I am using.

At the moment, CUCM 7.0 is using Web Config via the Web Browser. 

Can you let me know how to configure on CUCM 7.0?

I will appreciate very much if you give some instructions for
configuring SIP Trunk on CUCM7.0

Thanks
Kind regards,

Steven,

On 10 January 2011 19:33, Bogdan-Andrei Iancu <bogdan at voice-system.ro>
wrote:

Hi Steven,

To do that, you need to add in opensips some routing to 1) recognize the
numbers that needs to be sent to CUCM and 2)route that calls to CUCM.

For script logic it sounds like : if you receive a new call (initial
INVITE) for your local domain, check the URI and divert. If you look at
the default config file, there is comment "# requests for my domain" ->
from that point further you have only initial INVITEs for your local
domain, so you can add after:

  # all numbers starting with 55 are to be sent to CUCM
  if ($rU =~ "^55[0-9]+$") {
        # replace the domain part of RURI to point to CUCM
        rewritehostport("CUCM_IP:CUCM_PORT");
        # route the call out based on RURI
        route(1);
  }

For the other way around, you have to put a similar logic in CUCM, like
to divert all calls starting with "12" to opensips - and replace the
domain on RURI with the IP/domain of opensips.

Regards,
Bogdan

steven chew wrote:

	Hi Bogdan,

	Thank you very much for your reply.

	I have an Opensips Server and a Cisco Unified Communication
Manager (CUCM).

	If I want to make calls from Opensips Server to CUCM and CUCM to
Opensips Server.

	For example:
	1) If I dial an extension number "5566" from a SIP Phone "12345"
under Opensips Server, it will try to call to a Cisco IP Phone "5566"
from CUCM through a SIP Trunk.
	2) If I dial an extension number "12345" from a Cisco IP Phone
"5566" under CUCM, it will try to call to a SIP Phone "12345" under
Opensips Server through a SIP Trunk.

	Can you give some instructions how to configure the above
scenario for dialing extension numbers?

	Thanks
	Steven, 

	On 6 January 2011 21:31, Bogdan-Andrei Iancu
<bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:

	   Hi Steven,

	   If you use the opensips default script, your opensips will
accept
	   calls from any other external SIP entities (call targeting a
local
	   opensips subscriber).

	   If you want to configure your opensips to accept foreign
calls
	   only form a specific IP address, you can use the permission
	   module, with address table to implement IP-based
authentication.

	   Best regards,
	   Bogdan

	   steven chew wrote:

	       Hi everyone,

	       I am a newbie with SIP-Trunk in OpenSips.
	       I have a Cisco Communication Unified Manager and a
OpenSips
	       Server running in two different Virtual Machines.

	       I would like to have a SIP trunk in between them "Cisco
	       Communication Unified Manager and OpenSips Server".
	       Therefore, I can make a call from OpenSips Server's SIP
	       Clients to Cisco IP Phone.
	       What should I need to add into opensips.cfg configuration
file?

	       I hope you can give some simple examples how to do it.
	       I look forward to hearing from your advise asap.

	       Thanks
	       Regards,
	       -Steven.

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	   --     Bogdan-Andrei Iancu
	   OpenSIPS Event - expo, conf, social, bootcamp
	   2 - 4 February 2011, ITExpo, Miami,  USA

	   www.voice-system.ro <http://www.voice-system.ro>

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-- 
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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro

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