Closed GoogleCodeExporter closed 8 years ago
Very strange because this have been tested several times. There is an open
thread about DTMF
(https://groups.google.com/group/doubango/browse_thread/thread/c36164548bc12d41)
but not the same issue.
Are you using iDoubs v2?
Please not that DTMF digits will only sent when the RTP channel is UP which
means that you already accepted the call or early media started.
Did you already used wireshark and noticed there is no "rtpevent" packets
moving from the client?
Original comment by boss...@yahoo.fr
on 15 Jul 2011 at 4:01
is it applicable when we are making call?
Original comment by palejasu...@gmail.com
on 16 Jul 2011 at 7:47
Oke after 2,5 hours of extensively testing the SIP server I had to come to the
conclusion that it is an issue in the Doubango framework. The RTP Event is sent
to the server but the server doesn't receive anything.
I've also run whireshark on the SIP server but there is nothing to be found of
a RTP package that's coming trough. But when I try it with another SIP client
it works perfectly.
Original comment by nightfox...@gmail.com
on 16 Jul 2011 at 9:54
Attachments:
"the conclusion that it is an issue in the Doubango framework" => How could it
be an issue in Doubango if the packet are sent from the client and your server
doesn't receive anything? I would say it's an issue with your network unless
you find better explanation. I strictly follow RFC 4733 and already made many
IOTs.
The "rtpevent" packets use the same RTP channel as the voice so I don't see how
it could be possible to receive audio bu not dtmf digits.
Original comment by boss...@yahoo.fr
on 17 Jul 2011 at 12:33
[deleted comment]
I have tried DTMF Test. It is working properly when I am sending single digit,
Let us say 1. But what I have to do when I want to transfer the call to the
extesion number let us say 125.
I have tried to send it digit by digit, but call is transferd to first digit
only, or terminated.
It would be helpful if I will get code for it.
Original comment by palejasu...@gmail.com
on 18 Jul 2011 at 10:10
[deleted comment]
Hi Guys,
Great Work..!!
I agree to the Above..
DTMF works but only for Singal Digit.
E.g for an IVR where pressing Single Digit is required It works fine.
But When Multiple Digit e.g. 12,2333,etc.. is required the call gets
terminated.
The same issue is found IMSDROID & IDOUBS.
I am using Asterisk as my Testing Server& I have used X-Lite Phone as a SIP
client check the server conf & it works fine. I guess then their no problem
with Server & RTP.
Best of luck..!!
Original comment by narij...@gmail.com
on 18 Jul 2011 at 12:14
To all:
I have already said it and I'll say it again: All is correct with our
implementation and please check that the "rtpevent" packets send from Doubango
are not too short (duration) or long for your server. Most of open source
servers have it "hard coded".
Please don't say something like: "I have tested with xxx client and it's
working so the problem comes from iDoubs/IMSDroid/Boghe...."
If you found that there is a problem on our implementation (RFC 4733) please
point it and we will make the changes.
There is an discussion about DTMF issue here:
https://groups.google.com/group/doubango/browse_thread/thread/c36164548bc12d41
@narij...@gmail.com
It's not because it's working with x-lite that there is no problem on your
server.
Original comment by boss...@yahoo.fr
on 18 Jul 2011 at 4:28
Mamadou,
You owe me a sincere apology! :-)
After all we have discovered that the issue was the dtmfmode on the server
which was inband mode. After changing it to the rfc2833 mode it worked
perfectly.
I've never said anything :-#
Thanks for the feed back.
Original comment by nightfox...@gmail.com
on 19 Jul 2011 at 10:01
Hi Mamadou,
My Heartly apologies....!!
I had no intentions of showing or coming to a conclusion that it not works or
it works with Application X & not with iDoubs & IMSDroid so it has a bugs.. I
had no intentions of that.. Once again sorry for putting my concern in a wrong
frame of words..All i wanted is to mention that I had tested it with a
softphone and basic configuration..!!
After doing some Wireshark & other Research.. I Found out that am getting DTMF
three time.. ... e.g. if I am sending “1” I am getting “111” on my
server.. if i m pressing “23” I am getting “222333” so its sending each
digit three times..
Initially i thought it might be some issues with my Server so I re-installed
FreePBX but still getting the it three times.....
Once again Thanks for Giving us a awesome Framework & Application ... along
with a Promt support...
Regards
Original comment by narij...@gmail.com
on 19 Jul 2011 at 2:13
@narij...@gmail.com
Sorry for the rude response.
The retransmission is correct according to RFC 4733. For example, when you
press "2" you will have:
"2" -> Start
"2"
"2"
"2"
"2" -> End
"2" -> End
The retransmission is used to deal with packet lost or simulate "long press".
The retransmission is totally detectable because of the "Start" and "End". All
servers I've seen correctly handle this basic scenario. Are you developing your
own server?
Original comment by boss...@yahoo.fr
on 20 Jul 2011 at 2:25
Hi Mamadou,
Thanks for your reply.... & it’s absolutely fine I understand one can get
irritated by this kind of comments from a beginner like me for hard worked
project and trust me after seeing all your hard-work for development &
Support.. you should..!! ...Hope the misunderstanding is cleared..!!!
I am using a Asterisk 1.6.2.6... After Reading your comments I went through the
RTP debug on Asterisk server. Please find attached the RTP for RFC2388 I
received from Idoubs & X-Lite... there is some difference the way it comes to
the server. It would be great if you can guide what I am missing in my settings.
Regards..!!
Original comment by narij...@gmail.com
on 20 Jul 2011 at 1:26
Attachments:
What are the settings in the config file on your server? And does the server
recieve any of the packages at all?
What fixed the problem for me was to use one codec for example PCMA and set the
correct configuration on the server for PCMA dtmfmode (look for it on google)
there it'll be explained what type of dtmfmode you'll need to use.
Original comment by nightfox...@gmail.com
on 21 Jul 2011 at 1:28
Hi Guys,
@nightfox..
Thanks for your advice.. I googled arround to tweak settings in Asterisk.. and
it worked..
All i did is commented Following line in sip.conf of asterisk
dtmfrelax=yes
@Mamadou
Thanks for the help .. and no Hard Feelings. Once again thanks for amazing
stuff..
Cheers..!!
Original comment by narij...@gmail.com
on 29 Jul 2011 at 9:51
Original comment by boss...@yahoo.fr
on 8 Jun 2012 at 11:28
Original issue reported on code.google.com by
nightfox...@gmail.com
on 15 Jul 2011 at 9:40