[Freeswitch-users] How to limit call attemp time?
Code Ghar
codeghar at gmail.com
Sun Jun 13 14:26:02 PDT 2010
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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