I put on hold RFC2833 to try SIP INFO
but I got a 500 error from FS :
------------------------------------------------------------------------
INFO sip:8519999999999@192.168.0.1:5080;transport=udp SIP/2.0
Content-Length: 22
Via: SIP/2.0/UDP
192.168.0.1:30082;rport;branch=z9hG4bKxEcVSs7CyRX5jWOWKwkAeQ..
From: "FRANCK CHIONNA"
<sip:8511234567890@domain.com>;tag=93605822290
To:
<sip:8519999999999@domaine.com>;tag=7ZQ8v6jvSpmZm
Contact:
<sip:8511234567890@192.168.0.1:30082>
CSeq: 3
INFO
Max-Forwards: 70
Call-ID: 1279221367@192.168.0.1
Content-Type:
application/dtmf-relay
Signal=1
Duration=400
------------------------------------------------------------------------
tport_deliver(0x183cff0):
msg 0x7f099c06fdf0 (476 bytes) from udp/192.168.0.1:5080/sip next=(nil)
nta:
received INFO sip:8519999999999@192.168.0.1:5080;transport=udp SIP/2.0 (CSeq
3)
nta: INFO (3) going to existing INFO transaction
nta: timer I fired, terminate 407
response
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/3 free
nta:
timer I fired, terminate 200 response
nta_incoming_timer: 0/0 resent, 0/0
tout, 1/1 term, 1/2 free
tport_wakeup_pri(0x183cff0): events
IN
tport_recv_event(0x183cff0)
tport_recv_iovec(0x183cff0) msg
0x7f099c06fdf0 from (udp/192.168.0.1:5080) has 476 bytes, veclen = 1
recv 476
bytes from udp/[192.168.0.1]:30082 at 23:15:15.203315:
is the SIP message malformatted ?
can't see why it fails
Thanks
Franck
----- Original Message -----
Sent: Friday, May 07, 2010 6:04 PM
Subject: Re: [Freeswitch-users] Fw:
rfc2833 confusion
On Thu, May 6, 2010 at 12:23 PM, Madovsky
<infos@madovsky.org> wrote:
right,
but in case of the SIP caller needs to send digits to
FS,
his SIP phone should in any way send DTMF, isn't it
?
in fact I'm trying to implement rfc2833 in my own sip
phone through RTP,
but if there is a way to send from caller a string var
representing the digits typed to FS once bridge is done
and transfer it as dtmf digits I will be happy too...
Hook up a phone to your FS. If you have the default dialplan then
call 9999. Go to fs_cli and make sure you are at /log level 7
From
the phone press some digits. You will see them pop up on the display:
2010-05-07 15:03:56.407327 [DEBUG]
switch_rtp.c:2805 RTP RECV DTMF 1:1920
2010-05-07 15:03:56.407327 [DEBUG]
mod_dptools.c:1409 Digit 1
2010-05-07 15:03:57.688237 [DEBUG]
switch_rtp.c:2805 RTP RECV DTMF 2:2880
2010-05-07 15:03:57.688237 [DEBUG]
mod_dptools.c:1409 Digit 2
2010-05-07 15:03:58.669178 [DEBUG]
switch_rtp.c:2805 RTP RECV DTMF 5:3200
2010-05-07 15:03:58.669178 [DEBUG]
mod_dptools.c:1409 Digit 5
FreeSWITCH knows to look for
the DTMFs.
-MC
----- Original Message -----
Sent: Thursday, May 06, 2010 3:06
PM
Subject: Re: [Freeswitch-users] Fw:
rfc2833 confusion
Yes that allows you to play a prompt and then wait for the
person on the line to enter digits.
On Thu, May 6, 2010 at 2:15 PM, Madovsky
<infos@madovsky.org> wrote:
is this the right link ?
----- Original Message -----
Sent: Thursday, May 06, 2010 2:10 PM
Subject: Re: [Freeswitch-users] rfc2833
confusion
But how to catch digits form a dialplan when the
call
is already on ?
----- Original Message -----
Sent: Thursday, May 06, 2010 1:43
PM
Subject: Re: [Freeswitch-users]
rfc2833 confusion
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_dtmf
Don't
get to hung up on the inband / out of band thing. This will be
negotiated by FS and the carrier when the call is set up. For
instance, when you use G729 you need out of band DTMF but for G711 you
can use both. This might change from call you call, depending on what
media gateways can handle.
If you want to "see" and hear DTMF -
fire up wireshark (or Tshark) and capture the trace. You can see the
RTP DTMF in the trace and hear the clamped audio from the carrier
perhaps. Or just hear the inband tones. It gives a good visual of what
is going on.
On Thu, May 6, 2010 at 1:25 PM, Madovsky
<infos@madovsky.org> wrote:
I anyone knows a link of a programming example of
how
to inject rfc2833 in RTP flow (doesn't matter the
language)
It would be very useful
Thanks
F
----- Original Message -----
Sent: Thursday, May 06, 2010
12:39 PM
Subject: Re:
[Freeswitch-users] rfc2833 confusion
god, so many confusion for only an audio tone !
;)
anyway if I understand it's called inband by the
fact that it's in RTP flow onlly.
so the rfc2833 packet is only an event payload
...
----- Original Message -----
Sent: Thursday, May 06,
2010 11:52 AM
Subject: Re:
[Freeswitch-users] rfc2833 confusion
both 2833 and actual tones in the audio stream
are often called in-band because the 2833 is just a transport
for the inband tones that is assumed to be reconstructed on the
other side. Some people refer to inband as the real tones
and 2833 as not inband because it's not the tones. So the
one you have to look out for is people who call 2833 inband and
do not clarify that it's inband packetized according the
RFC2833
The one that is not inband for sure is INFO (which has been
dropped from the standard in favor of some ridiculous
SUBSCRIBE based xml exchange)
Like all things in SIP and anything else that comes with an
RFC even if you know the answer you can't rely on trusting that
answer because the real world will do whatever it wants and
invalidate you. =D
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