[OpenSIPS-Users] OPENSIPS+ASTERISK Integration

ram talk2ram at gmail.com
Wed Jun 9 09:34:41 CEST 2010


depends on the design

you can have one point of authenticaion and transaction

Ram



On Tue, Jun 8, 2010 at 10:59 PM, Premalatha Kuppan
<premalatha at ngintech.com>wrote:

> Thanks a lot.
>
> I have one question. If i route the call to asterisk for IVR ( in my case
> ivr is to authenticate the user to access the system), who will have the
> control, meaning who will maintian all the transactions and dailog. Is it
> opensips/Asterisk ?
>
> Thanks,
> Prem
>
>   On Tue, Jun 8, 2010 at 9:50 AM, Gabriel Bermudez <elgabo81 at gmail.com>wrote:
>
>> 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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