[OpenSIPS-Users] OPENSIPS+ASTERISK Integration

Premalatha Kuppan premalatha at ngintech.com
Tue Jun 8 19:29:56 CEST 2010


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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> >>>> Users at lists.opensips.org
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> >>>
> >>>
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> >>>
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> >>
> >>
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> >
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