How to implement a SIP Trunk in between two SIP servers.

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How to implement a SIP Trunk in between two SIP servers.

steven chew
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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Re: How to implement a SIP Trunk in between two SIP servers.

Bogdan-Andrei Iancu
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.
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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


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Re: How to implement a SIP Trunk in between two SIP servers.

steven chew
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 <[hidden email]> 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.

------------------------------------------------------------------------

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 


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


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: How to implement a SIP Trunk in between two SIP servers.

Bogdan-Andrei Iancu
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 <[hidden email]
> <mailto:[hidden email]>> 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.
>
>         ------------------------------------------------------------------------
>
>         _______________________________________________
>         Users mailing list
>         [hidden email] <mailto:[hidden email]>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>          
>
>
>
>     --
>     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>
>
>
>     _______________________________________________
>     Users mailing list
>     [hidden email] <mailto:[hidden email]>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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


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Re: How to implement a SIP Trunk in between two SIP servers.

steven chew
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 <[hidden email]> 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 <[hidden email] <mailto:[hidden email]>> 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.

       ------------------------------------------------------------------------

       _______________________________________________
       Users mailing list
       [hidden email] <mailto:[hidden email]>

       http://lists.opensips.org/cgi-bin/mailman/listinfo/users
       


   --     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>


   _______________________________________________
   Users mailing list
   [hidden email] <mailto:[hidden email]>

   http://lists.opensips.org/cgi-bin/mailman/listinfo/users


------------------------------------------------------------------------

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 


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


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: How to implement a SIP Trunk in between two SIP servers.

Bogdan-Andrei Iancu
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 <[hidden email]
> <mailto:[hidden email]>> 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
>         <[hidden email] <mailto:[hidden email]>
>         <mailto:[hidden email]
>         <mailto:[hidden email]>>> 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.
>
>              
>          ------------------------------------------------------------------------
>
>                _______________________________________________
>                Users mailing list
>                [hidden email]
>         <mailto:[hidden email]>
>         <mailto:[hidden email]
>         <mailto:[hidden email]>>
>
>                http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>                
>
>
>            --     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>
>
>
>            _______________________________________________
>            Users mailing list
>            [hidden email] <mailto:[hidden email]>
>         <mailto:[hidden email]
>         <mailto:[hidden email]>>
>
>            http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>         ------------------------------------------------------------------------
>
>         _______________________________________________
>         Users mailing list
>         [hidden email] <mailto:[hidden email]>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>          
>
>
>
>     --
>     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>
>
>
>     _______________________________________________
>     Users mailing list
>     [hidden email] <mailto:[hidden email]>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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


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Users mailing list
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Re: How to implement a SIP Trunk in between twoSIP servers.

Leon Li
In reply to this post by steven chew

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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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 <[hidden email] <mailto:[hidden email]>> 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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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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Re: How to implement a SIP Trunk in between twoSIP servers.

ahmeddd
This post was updated on .
hallo,

I have the same configuration problem but in my case I am using two IP phones ( two different VMs) and a sip server installed in another VM. Simple calls between users are working but one I try to call 666 ( number of the IP phone) the call does not proceed.
( i try to trunk calls between the two VMs 666 and 9999 using opensips server(x.x.15.18)

This how my config file looks:

   # all numbers starting with 66  number are to be sent to test8s2 and test8s7
     if ($rU =~ "^66[0-9]+$") {
          # replace the domain part of RURI to point to test8s2 or test8s7
          rewritehostport("10.42.13.82:5060");
          # route the call out based on RURI
          route(1);
     }

   # all numbers starting with 99 number are to be sent to test8s2 and test8s7
     if ($rU =~ "^99[0-9]+$"){
          # replace the domain part of RURI to point to test8s2 or test8s7
          rewritehostport("10.42.13.87:5060");
          # route the call out based on RURI
          route(1);
     }


route[1] {

        xlog("following route 1 ::: forwarding according to URI");
        # forward according to uri
         forward();


        # for INVITEs enable some additional helper routes
        if (!t_relay()) {
                sl_reply_error();
        };
        exit;
}



can any one help me in configuring the call !!

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Re: How to implement a SIP Trunk in between twoSIP servers.

ahmeddd
can any one help me !
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