asterisk users - Dec 2005 - Asterisk Call Recording and SIP canreinvite

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

> 
> 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

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

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

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
> >>>>> _______________________________________________
> >>>>> --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
> > _______________________________________________
> > --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

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
> >>>>>>>
_______________________________________________
> >>>>>>> --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
> >>> _______________________________________________
> >>> --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