[Asterisk-Users] Asterisk Call Recording and SIP canreinvite

[Steve Totaro]

There may be a better way but off the top of my head this idea jumped
out.  It assumes that you know prior to making the call that you need to
record it and that you have phones capable of multiple lines.  

Setup a second line with a different entry in sip.conf with
canreinvite=no and use that line to make your calls.  

Other than that I see reference on the wiki to an H option in dial but
have never used it.  I think you will still need to know prior to
dialing whether you will want to record the call or not so you can dial
the exten that uses the H option.

If you get this to work, please post your results back to this thread.

"Re: Re: H option
by flobi on Monday 25 of July, 2005 [10:43:46]
why not just set canreinvite=yes and on the calls where you don't want
reinvite use the H option (if it actually does disable reinvite) or the
T or t which also disable reinvite. 

7960G Seems to need canreinvite=no as well.
by Anonymous on Friday 29 of October, 2004 [22:22:43]
Running P0S3-07-2-00.

Re: H option
by Anonymous on Monday 26 of July, 2004 [10:10:07]
(:confused:) Hmm... Now I started to wonder, if it's somehow possible to
override the canreinvite=no setting on per call basis. Anyone?

H option
by Anonymous on Saturday 10 of July, 2004 [04:15:13]
Asterisk will not reinvite if the H option is used in the Dial command."

http://www.voip-info.org/wiki-Asterisk+sip+canreinvite

Thanks,
Steve

> 
> I understand. But because the majority of calls are not to be
> recorded, I don't have a need to keep Asterisk in the media path all
> the time. That's why I'm wondering if you could dynamically keep it
> in the media path or not.
> 
> - Waldo
> 
> On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote:
> 
> > Well, then set canreinvite=no
> >
> >>
> >> If that's the case, is it possible to override the canreinvite
> >> attribute of a SIP peer in extensions.conf before a call is made or
> >> answered by that peer?
> >>
> >> - Waldo
> >>
> >> On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote:
> >>
> >>>>
> >>>> Is there a way to optionally keep asterisk in the media path in
> > order
> >>>> to record calls using the Monitor command? For example, if I have
a
> >>>> SIP peer that is defined with canreinvite=yes, this means that if
> >>>> possible, Asterisk will not be in the media path. However, what
> >>>> happens if the user presses something like *1 (defined in
> >>>> features.conf) to record the call? Will the call be forced to go
> >>>> through Asterisk automatically?
> >>>>
> >>>> Thanks,
> >>>> Waldo
> >>>
> >>>
> >>> I could be wrong but I am pretty sure that once the asterisk is
out
> > of
> >>> the media path then features like *1 will not work since asterisk
> >>> is not
> >>> able to listen for it.
> >>>
> >>> Thanks,
> >>> Steve
> >>> _______________________________________________
> >>> --Bandwidth and Colocation provided by Easynews.com --
> >>>
> >>> Asterisk-Users mailing list
> >>> To UNSUBSCRIBE or update options visit:
> >>>    http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >> _______________________________________________
> >> --Bandwidth and Colocation provided by Easynews.com --
> >>
> >> Asterisk-Users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-users
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users