[OpenSIPS-Users] OPENSIPS+ASTERISK Integration

[Gabriel Bermudez]

Hi,

I basically use the load_balancer module to dispatch to different
asterisk servers

on the main route block after handling auth, registrations, etc

if(db_is_user_in("Request-URI", "ccivr")) {
          xlog("The call will be redirect to calling card server");
          route(3);
}

# route for call handled by calling card servers
route[3] {
        # for INVITEs enable some additional helper routes
        if (is_method("INVITE")) {
                t_on_branch("2");
                t_on_reply("2");
                t_on_failure("1");
                if(client_nat_test("15")) {
                        nat_keepalive();
                }
        }

        # prepare the message for the IVR

        # select less loaded IVR
        if(!load_balance("1", "ccivr")) {
                xlog("No IVR available !!!");
                sl_send_reply("503", "Service Unavailable");
                exit;
        };

        if(!t_relay()) {
                sl_reply_error();
                exit;
        }

        exit;

}


Hope it helps

Regards,

2010/6/8 ram <talk2ram at gmail.com>:
> I forgot the link
>
> i did this work some time back
>
> Lost the link, google it
>
> Opensips+asterisk+a2b
>
> Ram
>
> On Mon, Jun 7, 2010 at 4:07 PM, Premalatha Kuppan <premalatha at ngintech.com>
> wrote:
>>
>> Can you pleae guide me how to do this ?
>>
>> On Mon, Jun 7, 2010 at 4:04 PM, Douglas Lane <doug at wd.co.za> wrote:
>>>
>>> Hi Premalatha,
>>>
>>> Perhaps have a look at SEMS for this.
>>>
>>> Thanks
>>> Doug
>>>
>>>
>>> On 2010/06/07 12:30 PM, Premalatha Kuppan wrote:
>>>
>>> Thanks Sebastian.
>>>
>>> I have followed up this link and tried extending the opensips.cfg file to
>>> route call to Asterisk.
>>>
>>> I doubt/not clear that after IVR (meaning when the user is authenticated
>>> through IVR) who will handle all the transactions and dialog (OPENSIPS or
>>> Asterisk ) ?
>>>
>>> I want Opensips to handle all the transactions.
>>>
>>> Any insight on this ?
>>>
>>> Thanks,
>>> Prem
>>>
>>> On Mon, Jun 7, 2010 at 3:15 PM, Schumann Sebastian
>>> <Sebastian.Schumann at t-com.sk> wrote:
>>>>
>>>> Hi Prem
>>>>
>>>> There is a good tutorial at
>>>> http://www.opensips.org/Resources/DocsTutAsterisk It does exactly what you
>>>> need I assume.
>>>>
>>>> For details in writing and extending basic configuration, you can find
>>>> also the linked documentation there.
>>>>
>>>> Best regards
>>>> Sebastian
>>>>
>>>> > -----Original Message-----
>>>> > From: users-bounces at lists.opensips.org [mailto:users-
>>>> > bounces at lists.opensips.org] On Behalf Of Premalatha Kuppan
>>>> > Sent: Monday, 07. June 2010 11:35
>>>> > To: OpenSIPS users mailling list
>>>> > Subject: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration
>>>> >
>>>> > Hi,
>>>> >
>>>> > Can anyone guide me in building the Opensips and Asterisk Integration.
>>>> >
>>>> > I want to Use OpenSIPS as SIP PROXY (i.e all transactions and dialogs
>>>> > should be
>>>> > handled by Opensips) and Asterisk to do only IVR functionality.
>>>> >
>>>> > I appreciate if anyone can guide me in writing a routing logic from
>>>> > Opensips to
>>>> > Asterisk for IVR and futher call flow OPENSIPS to handle.
>>>> >
>>>> > Thanks,
>>>> > Prem
>>>> >
>>>>
>>>>
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>>>
>>>
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>>
>>
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>
>
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