[Asterisk-Users] Asterisk Call Recording and SIP canreinvite

[Steve Totaro]

Do you have canreinvite=yes?  If you do change it to no.  If that works
then read the rest of this thread for options if you do not want all
streams to through asterisk.

Thanks,
Steve

> 
> I have a related issue.
> 
> I have everything set up correctly so that I CAN use live recording
> (Press *1 to start and stop recording.)
> When I press *1, the console indicates "user pressed *1 to start
> recording."  I also hear the "beep" and an audio file is created.
> The problem is that the audio file IS NOTHING BUT SILENCE.  It is the
> correct length, but only contains silence.
> 
> Any ideas???
> 
> -N
> 
> 
> On Dec 8, 2005, at 8:49 AM, Steve Totaro wrote:
> 
> > Yeah, makes sense now that I think about it a little more.  Guess
you
> > will have to prefix your exten so that the dial string with the H is
> > used and dial that prefix when you know or think that you may have
to
> > record a call.
> >
> >>
> >> This and Time Bandit's comment makes sense. I didn't realize that
> >> these options in the Dial string will "force" Asterisk to stay in
the
> >> media path even if canreinvite=yes.
> >>
> >> I'll give it a try.
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote:
> >>
> >>> 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
> >>>>>>> _______________________________________________
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