[OpenSIPS-Users] Generate INFO application/dtmf-relay message

Mon Nov 24 20:39:46 CET 2008

Still, regardless of the specific requirements. You've mentioned that it's a
analog phone.. And thus you won't have a connected line to it without
"calling" it.
Maybe you could tell us more about it and we'd  be able to give you some
idea of how you might be able to do it?
-Brett

On Mon, Nov 24, 2008 at 12:56 PM, Giuseppe Roberti <jnod at jnod.org> wrote:

> Hi brett.
>
> I really would be able to re-think what i have to do.
> But the device is not on my control and i need to satisfy my customer.
> However i don't need to speak with this device before the call is
> started but immediately after the caller response.
> And i can't do it from the caller.
> When i have read about SIP INFO DTMF i have decided to try if it can
> work for me, but its not so, because a sip proxy cant generate transaction.
>
> The only think i can try now is to inject RTP.
> I use mediaproxy for every call that come from those devices, so i see
> two possibility.
> The first, send RTP to media-relay caller address/port (and maybe i also
> need to send it from ip/port of the caller, using raw socket) so it than
> be relayed to the callee.
> The second is to send directly to callee address/port audio rtp from
> media-relay server, using the same address/port used by the relay for
> this call audio stream, but if so i think i have to do it before the
> port become used by conntrack, isnt it ?
> Please send me any comment/suggestion.
>
> Regards.
>
> Brett Nemeroff wrote:
> > I really, think you need to re-think what you are trying to do. It simply
> > doesn't make any sense the scenario you've described.
> > I imagine your mystery device is something like a door control. If it's
> > analog, and you are using some sort of ATA, it doesn't make sense to send
> it
> > a DTMF before you make a call to it.. That would be like leaving a
> message
> > on your home answering machine without ever calling it. If your analog
> > mystery device requires a telephone line, you'll need to place a call to
> it
> > first.. Once you have the dialog established, it should be pretty easy to
> > send it a tone from whatever established the dialog.
> > -Brett
> >
> >
> > On Sun, Nov 23, 2008 at 9:10 AM, Giuseppe Roberti <jnod at jnod.org> wrote:
> >
> >> Hi Bogdan.
> >>
> >> Sorry, i cannot reply to your request.
> >> Simply i don't know if the INFO will be a in-dialog request or not.
> >> I am learning sip and opensips day by day.
> >>
> >> Thank you, i will try t_uac_dlg()
> >>
> >> Regards.
> >>
> >> Bogdan-Andrei Iancu wrote:
> >>> Hi Brett,
> >>>
> >>> I think you refer to the TM module (not UAC) for generating SIP
> requests
> >>> - the t_uac_dlg MI command:
> >>>     http://www.opensips.org/html/docs/modules/1.4.x/tm.html#id2493975
> >>>
> >>> As Giuseppe mentioned, the DTMFs he needs are transported via SIP INFO
> >>> method, so OpenSIPS could generate such a method. The problem is
> (please
> >>> correct me if I'm wrong) that the INFO will a in-dialog request (INFO
> >>> request will belong to the INVITE dialog). And OpenSIPS cannot generate
> >>> within the dialog request (because of the old Cseq story).
> >>>
> >>> Regards,
> >>> Bogdan
> >>>
> >>> Brett Nemeroff wrote:
> >>>> I've mentioned several times.. try using the UAC module to generate
> >>>> the message. OpenSIPs is a proxy. It isn't made to GENERATE any
> >>>> message, but to relay and reply to messages. You can use the fifo
> >>>> method of the UAC module to generate an outgoing message. Look up the
> >>>> click to dial (ctd) examples. It's not for SIP INFO, but it has an
> >>>> INVITE method.
> >>>>
> >>>> On the other hand:
> >>>> sipp (or some sort of real UAC)----> opensips --->
> >>>> your-fancy-device-we-don't-understand
> >>>> makes plenty of sense with the given technology
> >>>>
> >>>> That's my $0.02
> >>>>
> >>>> On Mon, Nov 17, 2008 at 10:41 PM, Giuseppe Roberti <jnod99 at gmail.com>
> >> wrote:
> >>>>> I need only to send an INFO request that describe a DTMF tone, an
> INFO
> >>>>> application/dtmf-relay request. I dont need sipp nor any other media
> >>>>> software.
> >>>>> Take a look to http://www.voip-info.org/wiki/view/SIP+Info+DTMF
> >>>>> Its only a think that is relative to uac instead of proxy.
> >>>>> Maybe its better to talk about on other places.
> >>>>>
> >>>>> Regards.
> >>>>>
> >>>>> Brett Nemeroff wrote:
> >>>>>
> >>>>>> Perhaps you can use sipp to generate a call and send RTP
> >>>>>>
> >>>>>> On Mon, Nov 17, 2008 at 7:24 PM, Giuseppe Roberti <jnod at jnod.org>
> >> wrote:
> >>>>>>> Ill try.
> >>>>>>>
> >>>>>>> I have a custom hardware phone that do something when receive a
> dtmf
> >> (it
> >>>>>>> cant read sip, its an analogical phone).
> >>>>>>> So i have to send dtmf when some event occur (like, received new
> >> invite,
> >>>>>>> 200 ok for invite, etc...) to a sip ata that encode it for the
> >>>>>>> analogical hardware.
> >>>>>>> I know that a proxy does not generate request but i personally need
> >> it.
> >>>>>>> Thanks again.
> >>>>>>>
> >>>>>>> brett at nemeroff.com wrote:
> >>>>>>>
> >>>>>>>> What about using the uac module and generating it manually, like
> how
> >> the ctd example works??
> >>>>>>>> I'm not sure why you'd want to do this....
> >>>>>>>> Sent from my Verizon Wireless BlackBerry
> >>>>>>>>
> >>>>>>>> -----Original Message-----
> >>>>>>>> From: Alex Balashov <abalashov at evaristesys.com>
> >>>>>>>>
> >>>>>>>> Date: Mon, 17 Nov 2008 19:25:31
> >>>>>>>> To: Giuseppe Roberti<jnod at jnod.org>
> >>>>>>>> Cc: <users at lists.opensips.org>
> >>>>>>>> Subject: Re: [OpenSIPS-Users] Generate INFO application/dtmf-relay
> >> message
> >>>>>>>>
> >>>>>>>> The fact remains that a proxy cannot originate requests.  They
> would
> >>>>>>>> have to come from another UAC.
> >>>>>>>>
> >>>>>>>> Giuseppe Roberti wrote:
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>> Hi.
> >>>>>>>>>
> >>>>>>>>> Sorry. I'll try to explain better.
> >>>>>>>>> I want to send an INFO request from the UAS to one UAC.
> >>>>>>>>> This request is a SIP request, specified by rfc 2976, it's not
> RTP
> >> media.
> >>>>>>>>> It's like this: http://www.voip-info.org/wiki/view/SIP+Info+DTMF
> >>>>>>>>>
> >>>>>>>>> Thank you all.
> >>>>>>>>>
> >>>>>>>>> Bogdan-Andrei Iancu wrote:
> >>>>>>>>>
> >>>>>>>>>> Hi Giuseppe,
> >>>>>>>>>>
> >>>>>>>>>> OpenSIPS is a SIP proxy and has no media related capabilities,
> so
> >> it is
> >>>>>>>>>> not able to generate DTMF tones.
> >>>>>>>>>>
> >>>>>>>>>> Regards,
> >>>>>>>>>> Bogdan
> >>>>>>>>>>
> >>>>>>>>>> Giuseppe Roberti wrote:
> >>>>>>>>>>
> >>>>>>>>>>> Hi.
> >>>>>>>>>>>
> >>>>>>>>>>> I would able to generate a DTMF from opensips.
> >>>>>>>>>>> Is it possible ? How can i do it ?
> >>>>>>>>>>>
> >>>>>>>>>>> Regards.
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>> --
> >>>>>>> Giuseppe Roberti
> >>>>>>> <jnod at jnod.org>
> >>>>>>>
> >>>>>>>
> >>>>> --
> >>>>> Giuseppe Roberti
> >>>>> <jnod at jnod.org>
> >>>>>
> >>>>>
> >>>> _______________________________________________
> >>>> Users mailing list
> >>>> Users at lists.opensips.org
> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> Users mailing list
> >>> Users at lists.opensips.org
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >> --
> >> Giuseppe Roberti
> >> <jnod at jnod.org>
> >>
> >
>
>
> --
> Giuseppe Roberti
> <jnod at jnod.org>
>
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