Summary:ASTERISK-00113: SIP rfc2833 DTMF eaten before reaching SIP extension.
Reporter:yamez (yamez)Labels:
Date Opened:2003-08-17 22:00:44Date Closed:2011-06-07 14:10:30
Versions:Frequency of
Description:When making a sip to sip call through Asterisk with
canreinvite=no and dtmfmode=rfc2833 on each extension
no tone is heard on the remote phone.
Comments:By: Brian West (bkw918) 2003-08-17 22:37:57

sip to sip why would tone need to be preset on the other end of the call?  Also what sip devices are you using?

By: Mark Spencer (markster) 2003-08-23 13:47:54

Is there still any interest in this ticket?

By: Brian West (bkw918) 2003-08-23 14:53:50

I think it can be closed.  Because the remote sip device should know what to do with the RFC2833 tones. If it doesnt then the sip device is at fault.  Or you can pass DTMF inband to solve this.

By: yamez (yamez) 2003-08-25 14:49:37

This worked about two weeks ago, Setting it to inband does mostly fix the problem,
(The tone are still to short)  But inband is not gonna work for lose-full codecs.
This is an asterisk problem. If you have to cisco or budgetone phone and you call one from the other with canreinvite=no in the sip profiles you can bang on the keypad all day long and not hear a thing on the far end.   rfc2833 and info might as well be removed from asterisk at this point because there mostly usless.

By: yamez (yamez) 2003-08-25 14:58:50

Markster, How does one show intrest in a ticket?  I sit in IRC and wait for you to come in... I have yet to see you online. <grin>  What is a good time to hang out in #asterisk?  Durring biz hours I check in once every few hours or so for a few minutes I'm afraid I don't have a job such that I can sit and chill in IRC all day <smile>
But I would love one. Know of any? I did send a few emails to your sales@digium.com address requesting info about some kind for-pay support contracts, But have not got any-e-thing back yet. I can work my hours around to over lap with yours if I just know what they are.

By: yamez (yamez) 2003-08-25 15:01:12

To: bkw918:  "sip to sip why would tone need to be preset on the other end of the call? Also what sip devices are you using?" IVR apps?

By: Brian West (bkw918) 2003-08-25 15:02:31

what IRC network are you on?  Not irc.freenode.net.. because kram is on their all the time.

By: Mark Spencer (markster) 2003-08-25 15:24:42

Obviously we use both INFO and rfc2833 all the time, so the question is why it isn't working in your particular environment.

Can you call IVR on Asterisk and operate with the phone in rfc2833 mode?  If so, then clearly we *are* receiving rfc2833.

I'm on irc.freenode.net in #asterisk as kram.

By: yamez (yamez) 2003-08-25 15:52:13

Yes IVR on Asterisk does work (yes it does get and use DTMF via rfc2833)
but Asterisk can not _send_ or _bridge_ tones.

cisco 7960(rfc2833)(5.3)  -->  Asterisk(latest CVS) --> cisco 7960 (rfc2833)(5.3)
You get no tones through.

cisco7960(rfc2833) ---> Asterisk ----> cisco 5300 ---> pots
No tones

cisco7960(inband) -----> Asterisk -----> cisco 53000 ---->pots
Get very short tones (most IVRs just ignore them)

budgetone(info) ---->  Asterisk ----> vocal ---->  pots
no tones

Take Asterisk out of the loop and any of these and they work.

By: yamez (yamez) 2003-08-29 08:51:47

The work around in bug http://bugs.digium.com/bug_view_page.php?bug_id=0000171
fixes the problem with  SIP rfc2833 DTMF eaten before reaching the SIP gateway.

It does not fix the problem (this ticket is open for) with  
SIPphone ---> asterisk ----> SIPphone
when both phone use the dtmfmode=rfc2833 in sip.conf,
As the gatway problem is the most common I though I would post this work around here.

By: John Todd (jtodd) 2003-09-08 18:41:20

Yamez: yes, that's correct.  RFC2833 is not a tone, but an in-band signal.  There is no tone generated by Asterisk in this case.  Your SIP phone receives the RFC2833 message, and then the PHONE must play the touch tone.  In all of your examples, you do not say if you have ever heard tones with this configuration:

Cisco 7960(RFC2833)  ->  Cisco 7960(RFC2833)

This will not work, with or without Asterisk in the middle.  No tones will be played, since that is not how SIP phones work.

None of your examples show Asterisk as a POTs termination, either, where it would work just fine since RFC2833 would be converted into DTMF by Asterisk, and not by the "other" edge device.  I have had difficulty getting Cisco PSTN (3640 with VoIP cards) devices to correctly handle DTMF using RFC2833, but this is clearly not an Asterisk problem.  Does this clarify anything, or do you still think this is an Asterisk problem?

By: John Todd (jtodd) 2003-10-16 04:34:57

Checking up on this ticket; does my explanation make sense, or am I missing what you're trying to say?

By: yamez (yamez) 2003-10-24 06:44:03

Not really as rfc2833 is an out of band signal. <smile>

I have tested this with out Asterisk.
Phone --> Vocal --> Phone works.
Also Phone --> Cisco 5300 --> PSTN also works.  
The problem seem to be SIPphone --> Asterisk --> SIPphone.
I have dtmfmode=rfc2833  in each and every SIP profile in sip.conf
I assume this means that no matter what type of tone Asterisk gets via
other channles (inband, info or rfc2833) that only rfc2833 will be sent to my phones?  They have no problems getting rfc2833 from my Cisco 5300 if I drop Asterisk out of the picture. Or oddly enough if the phones send info or inband instead of rfc2833.  I could have assumed wrong?

Like I said I have worked around the problem on my end. But I do think it is a bug and an Asterisk bug at that. Asterisk will not send info or rfc2833 correctly to sip phones registered to it if it get said message via another sip device. I'm sure it work great as long as Asterisk is the PSTN gateway, but not everyone uses Asterisk as a PSTN gateway. (Although I might soon as the sip code seems to be getting a bit more stable as of late.)