Hi Steven,

sorry, but I know nothing on CUCM :(

Regards,
Bogdan

steven chew wrote:
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 <[email protected] <mailto:[email protected]>> 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
        <[email protected] <mailto:[email protected]>
        <mailto:[email protected]
        <mailto:[email protected]>>> 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>
        <http://www.voice-system.ro>


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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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