[Freeswitch-users] How to limit call attemp time?

Sergey Okhapkin sos at sokhapkin.dyndns.org
Sun Jun 13 15:02:03 PDT 2010 

However I think it's FS bug. call_timeout should wait for answer, regardless 
of early media.

On Sunday 13 June 2010, Sergey Okhapkin wrote:
> Thanks! bypass_media=true in combination with ignore_early_media=true did
>  the trick.
> 
> On Sunday 13 June 2010, Code Ghar wrote:
> > I tested it the following way:
> >
> > <extension name="outside-call">
> > <condition field="destination_number" expression="^2125550000$">
> > <action application="set" data="continue_on_fail=true" />
> > <action application="set" data="ignore_early_media=true" />
> > <action application="set" data="call_timeout=10" />
> > <action application="bridge" data="sofia/gateway/gateway1/$1"/>
> > <action application="sleep" data="5000" />
> > <action application="set" data="call_timeout=15" />
> > <action application="bridge" data="sofia/gateway/gateway2/$1" />
> > <action application="set" data="call_timeout=30" />
> > <action application="bridge" data="sofia/gateway/gateway3/$1" />
> > </condition>
> > </extension>
> >
> > After 10 seconds, FS sends "cancel" to gateway1 and sends "invite" to
> > gateway2 after sleeping for a while (optional). Once it sends invite to
> > gateway2, after 15 seconds FS sends "cancel" to gateway2 and sends
> > "invite" to gateway3. If after 30 seconds call is not answered it sends
> > "cancel" to gateway3 and then does not call any other gateway. From all
> > three gateways I received "183 Session Progress".
> >
> > If at any point call is answered, then subsequent gateways are not tried.
> > For example, if call is answered through gateway1, then after hangup it
> > doesn't attempt gateway2 and gateway3.
> >
> > Now comes your situation, where you need to try one gateway only. In this
> > situation the following may be of help.
> >
> > <extension name="outside-call">
> > <condition field="destination_number" expression="^2125550000$">
> > <action application="set" data="continue_on_fail=true" />
> > <action application="set" data="ignore_early_media=true" />
> > <action application="set" data="call_timeout=10" />
> > <action application="bridge" data="sofia/gateway/gateway1/$1"/>
> > </condition>
> > </extension>
> >
> > As you can see, we have eliminated all gateways except gateway1. I have
> > tested this myself and after 10 seconds FS sends "cancel" to gateway1 and
> > call is disconnected.
> >
> > Hope this helps. I used the examples at following wiki links to create
> > and test the above configurations.
> >
> > Misc. Dialplan Tools bridge
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
> >
> > Dialplan FollowMe
> > http://wiki.freeswitch.org/wiki/Dialplan_FollowMe
> >
> >
> >
> > On Sun, Jun 13, 2010 at 1:56 PM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > Just tried,  leg_timeout doesn't work too.
> > >
> > > On Sunday 13 June 2010, Peder wrote:
> > > > How about leg_timeout on that specific leg of the call.  I had an
> > > > issue where a general call_timeout didn't work if there was no
> > > > response but leg_timeout did work.  Not sure exactly how it works if
> > > > there is a
> > >
> > > response
> > >
> > > > though.  Worth a shot.
> > > >
> > > > -----Original Message-----
> > > > From: freeswitch-users-bounces at lists.freeswitch.org
> > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > >
> > > Sergey
> > >
> > > > Okhapkin
> > > > Sent: Sunday, June 13, 2010 1:25 PM
> > > > To: freeswitch-users at lists.freeswitch.org
> > > > Subject: Re: [Freeswitch-users] How to limit call attemp time?
> > > >
> > > > Are you getting SIP 180 or SIP 183 from the B leg? Everything works
> > > > fine when
> > > > SIP 180 is received, but call_timeout doesn't work when SIP 183 is
> > > >  received.
> > > >
> > > > switch_ivr_originate.c checks MEDIA READY condition, but not for
> > > > ANSWER condition, "media ready" includes early media also :-(
> > > >
> > > > On Sunday 13 June 2010, Madovsky wrote:
> > > > > Sergey,
> > > > >
> > > > > I did this
> > > > >
> > > > > <action application="set" data="continue_on_fail=true"/>
> > > > >   <action application="ring_ready"/>repk
> > > > >    <action application="set" data="call_timeout=20"/>
> > > > >
> > > > > before bridge and works for me
> > > > >
> > > > > Franck
> > > > >
> > > > > ----- Original Message -----
> > > > > From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
> > > > > To: <freeswitch-users at lists.freeswitch.org>
> > > > > Sent: Sunday, June 13, 2010 2:05 PM
> > > > > Subject: Re: [Freeswitch-users] How to limit call attemp time?
> > > > >
> > > > >
> > > > > Correct, I need to limit ringing time and continue dialplan
> > > > > execution
> > >
> > > if
> > >
> > > > no
> > > >
> > > > > answer within N seconds.
> > > > >
> > > > > It's trivial "find me" service, serial DID forwarding to multiple
> > > > > PSTN numbers.
> > > > >
> > > > > On Sunday 13 June 2010, David Ponzone wrote:
> > > > > > What you want is a way to limit the ringing time isnt it ?
> > > > > > I think it's not possible. I needed that some time ago, and I
> > > > > > never found out the solution.
> > > > > >
> > > > > > David Ponzone  Direction Technique
> > > > > > email: david.ponzone at ipeva.fr
> > > > > > tel:      01 74 03 18 97
> > > > > > gsm:   06 66 98 76 34
> > > > > >
> > > > > > Service Client IPeva
> > > > > > tel:      0811 46 26 26
> > > > > > www.ipeva.fr  -   www.ipeva-studio.com
> > > > > >
> > > > > > Ce message et toutes les pièces jointes sont confidentiels et
> > > > > > établis à l'intention exclusive de ses destinataires. Toute
> > > > > > utilisation ou diffusion non autorisée est interdite. Tout
> > > > > > message électronique est susceptible d'altération. IPeva décline
> > > > > > toute responsabilité au titre de ce message s'il a été altéré,
> > > > > > déformé ou falsifié. Si vous n'êtes pas destinataire de ce
> > > > > > message, merci de le détruire immédiatement et d'avertir
> > > > > > l'expéditeur.
> > > > > >
> > > > > > Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
> > > > > > > progress_timeout exits the bridge if NO 180 or 183 received,
> > > > > > > I'm receiving
> > > > > > > 183.
> > > > > > >
> > > > > > > On Sunday 13 June 2010, João Mesquita wrote:
> > > > > > >> Look at leg_progress_timeout or just progress_timeout.
> > > > > > >>
> > > > > > >> On Sunday, June 13, 2010, Sergey Okhapkin
> > > > > > >>
> > > > > > >> <sos at sokhapkin.dyndns.org> wrote:
> > > > > > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if
> > > > > > >>> early media
> > > > > > >>> received.
> > > > > > >>>
> > > > > > >>> On Sunday 13 June 2010, Code Ghar wrote:
> > > > > > >>>> Hi Sergey
> > > > > > >>>>
> > > > > > >>>> Have you tried ignore_early_media? It may help. You can find
> > >
> > > more
> > >
> > > > > > >>>> information from
> > >
> > > http://wiki.freeswitch.org/wiki/Channel_Variables
> > >
> > > > > > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
> > > > > > >>>>
> > > > > > >>>> <sos at sokhapkin.dyndns.org>wrote:
> > > > > > >>>>> Which variable should be set to exit bridge application if
> > > > > > >>>>> the call
> > > > > > >>>>> has not been answered within the specified time?
> > > > > > >>>>> call_timeout variable
> > > > > > >>>>> works if called
> > > > > > >>>>> SIP end point responds with "180 Ringing", but doesn't work
> > > > > > >>>>> if the endpoint responds with early media.
> > > > > > >>>>>
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