[Freeswitch-dev] Searching for a valid extension.

Thu Oct 14 10:16:54 PDT 2010

Thanks for the sugestions.

I checked carefully every step, however, I got some questions.

We tested the originate with the loopback endpoint, because we need to pass
through the dialplan.

My context:

<context name="khomp-SIP">
  <extension name="SIP">
    <condition field="destination_number" expression="^10(\d)$">
      <action application="bridge" data="user/100$1"/>
    </condition>
  </extension>
</context>

Test:
freeswitch at geovani> originate loopback/10/khomp-SIP/XML 1002

-ERR NO_ROUTE_DESTINATION

In this test, we send the digits 10 (simulating the reception of these
digits in the FXS) for khomp-SIP context in XML. I expected the
INVALID_NUMBER_FORMAT return (accept more digits and try again), but the
result was NO_ROUTE_DESTINATION. This happened with the switch_ivr_originate
function too.

switch_ivr_originate(NULL, &session, &cause, dialstring.c_str(), timeout,
NULL, NULL, NULL, NULL, NULL, SOF_NONE, NULL);

cause = 3 = NO_ROUTE_DESTINATION

In previous email I did not say, we can make calls with timeout between
digits, but my goal is to anticipate the call if possible checking of the
extension/context(exact match). Also, we need verify what context configured
by the user is used.

Thanks again Anthony Minessale and Mathieu Parent, but the others options
use timeout for connections, if we understand completely.

On Mon, Oct 11, 2010 at 5:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> We don't have a static monolithic dialplan like asterisk, there can be
> many dialplans or dynamic dialplans that reach out to external
> services.
>
> The proper way to implement overlap dialing on a per-protocol basis would
> be:
>
> 1) Separate your native channel from the FreeSWITCH channel so you can
> create a FreeSWITCH session and attach the native channel.
>
> 2) send the call to the configured dialplan and allow that dialplan
> module to work out it's own ambiguities and respond:
>
> SWITCH_CAUSE_INVALID_NUMBER_FORMAT or cause 28
>
> 3) when you see this cause accept more digits and try again until you
> get some other cause code.
>
>
> This is how it's done in sip by doing an entire new call each time you
> need to dial more digits.
>
>
>
> The other way which we do in FreeTDM is to define 2 regex, one for
> anything that is valid and one for anything that is invalid and
> consult this regex on each digit until you know one way or another.
>
>
>
>
>
>
> On Mon, Oct 11, 2010 at 3:12 PM, Geovani Ricardo Wiedenhoft
> <grw.freeswitch at gmail.com> wrote:
> > Hello,
> >
> > We are completing development of the mod_khomp endpoint for Khomp boards:
> > - FXS
> > - FXO
> > - E1
> > - GSM (boards and usb devices)
> > - Passive record
> > - kommuter
> >
> > However, we are constrained in our implementation. We needed to verify
> that
> > a given set of digits has a valid extension or if can wait for more
> digits
> > at some points of our code, without making the call.
> >
> > In our implementation for Asterisk (our Khomp channel), we use the
> functions
> > provided by the Asterisk:
> >
> > ##############################
> > ###############################################
> > ASTERISK:
> >
> > /* \brief Looks for a valid matching extension */
> >  - ast_canmatch_extension
> >
> > /* \brief Looks to see if adding anything to this extension might match
> > something. (exists ^ canmatch) */
> >  - ast_matchmore_extension
> >
> #############################################################################
> >
> > Searching for a solution, we implemented within mod_khomp the reading and
> > checking functions of digits and expressions in XML, but our
> implementation
> > limits us to XML dialplans, which it would not be compatible if the user
> > need the database or other constructions of the dialplan.
> >
> >
> > So, I would like to know what is the possibility of similar functions to
> be
> > implemented in the core?
> >
> >
> >
> >
> >
> >
> > Thanks
> > :)
> >
> >
> >
> >
> > Geovani Ricardo Wiedenhoft
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
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>
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>
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>
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