[asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

[SIP]

It's a valid concern, but be prepared for people to tell you that this
should be done with the qualify parameter to determine if a host is up
and running. Not the most ideal way to handle it, I'll agree. But the
SIP proxy functionality of Asterisk is limited (as it's not intended to
be a SIP proxy).

We use a modified source to enable the return of SIP response codes into
our AGI scripts that help us do more intelligent logic in this regard
(mostly for LCR functionality).  It could be modified, I suppose, to
look for the return of a provisional response indicating that the remote
proxy is up and running and responding to SIP requests. Not sure how
you'd then initiate a timeout, though.

N.


Julien Chavanton wrote:
>  
> "You want to know if the remote address/proxy is up and running before you
> bother trying to wait on it for very long. Is this right?" , yes this
> would be a good start ?
>  
> - But the IP could be up and the SIP service down, we need a signaling
> timeout, I beleive a good way in term of responsability would be :
>  If I do not receive a response to the SIP INVITE in timeout duration
> then I would cancel the call and try with another route.
>  
> - With AGI can we control and react to the signaling events, I guess not ?
> Thank you
>  
> ------------------------------------------------------------------------
> *From:* asterisk-users-bounces at lists.digium.com on behalf of SIP
> *Sent:* Thu 18/12/2008 6:13 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Dial timeout with SIP - how to set
> timeout for INVITE ACK
>
>
> >
> > ------------------------------------------------------------------------
> > *From:* asterisk-users-bounces at lists.digium.com on behalf of Philipp
> > Kempgen
> > *Sent:* Thu 18/12/2008 4:17 PM
> > *To:* Asterisk Users
> > *Subject:* Re: [asterisk-users] Dial timeout with SIP - how to set
> > timeout for INVITE ACK
> >
> > Julien Chavanton schrieb:
> > > I have a concern with Dial command, I want to enable a secondary
> > route with a remote partner, if the first route fails then we use the
> > second one :
> >
> > > Solution1: it will try both (there will be 2 simultanious actives
> > calls ringing) this is not clean when calling an endusers
> > >
> > >  exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5
> > <SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/$%7BEXTEN%7D at remote-sip1,5>> )
> > >  exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip2,5
> > <SIP/${EXTEN}@remote-sip2,5 <mailto:SIP/$%7BEXTEN%7D at remote-sip2,5>> )
> >
> > You can't have the same "priority" (1) more than once per
> > extension (_X.).
> >
> > > Solution2: it will wait until 5 seconds of timeout (on answer) and
> > then try the second alternative "n"
> > >
> > >  exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5
> > <SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/$%7BEXTEN%7D at remote-sip1,5>> )
> > >  exten => _X.,n,Dial(SIP/${EXTEN}@remote-sip2,5
> > <SIP/${EXTEN}@remote-sip2,5 <mailto:SIP/$%7BEXTEN%7D at remote-sip2,5>> )
> > >
> > > the problem is we can not select what timeout represents, timeout on
> > ACK from INVITE would be perfect I think (1 second for example),
> > timeout for answer ? this is to hard to predict, some mobile phone can
> > ring for 30 seconds, etc.
> >
> > So why not use 30 and let Asterisk take care of the SIP details/
> > timeouts?
> >
> > And just to be sure: Don't put those "mailto" things in
> > extensions.conf.  :-)
> >
> >
> >    Philipp Kempgen
> >
> Julien Chavanton wrote:
> > >So why not use 30 and let Asterisk take care of the SIP details/
> > >timeouts?
> > 
> > Asterisk will wait the until it receive "answer" or timeout
> > 
> > I need to timeout a SIP call on SIP INVITE ACK, in ISDN for exmaple
> > this is translated to PROCEEDING
> > Meaning "I have received the call, now I will look what to do with it"
> > 
> > The result with the suggested timeout is not good enought, you may
> > wait for the whole timeout even if the other side as not sent
> > anything, this will be the case for all your calls, depending on the
> > timeout this would be killing the traffic.
> > 
> > 
>
> It sounds as though you want the result of the SIP INVITE (looking for,
> say, a provisional 1XX response) and want the timeout to be set for
> whether or not you receive the provisional response in time?  i.e. You
> want to know if the remote address/proxy is up and running before you
> bother trying to wait on it for very long. Is this right? Or am I
> missing the point of the question?
>
> N.
>