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Incomming DTMF not working

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Mori

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I choose to make a new thread with this subject.

The 3CX is in general working perfectly.
My only problem so far is that receptionist does not register DTMF dials from incomming calls.

The original thread came from http://www.3cx.com/forums/viewtopic.php?t=1072
 
is this from inbound calls via a VOIP Gateway or VOIP provider?

Are you using G711 as the codec? Note that in band DTMF recognition can only work with G711. This is true for all IP phone systems. GSM will not work. nor will G729 (which we dont support anyway)

if it is from a local call, ensure that your phone is sending DTMF 'in band', not out of band. In band is the convential way, as a tone.
 
I'll try to answer my best.

admin said:
is this from inbound calls via a VOIP Gateway or VOIP provider?

Under "Lines" i registered a VoIP Provider, and the 3CX made a gateway from that.

Are you using G711 as the codec? Note that in band DTMF recognition can only work with G711. This is true for all IP phone systems. GSM will not work. nor will G729 (which we dont support anyway)

My 3CX stands on a win2k3 server directly behind an Belkin N1 WL Router, connected through Thompson Cable Modem to the Provider.
There is no option for me to change any codecs here. (With that i mean from the provider->modem->router->3CX)
All ports needed is opend in and out of the router..

if it is from a local call, ensure that your phone is sending DTMF 'in band', not out of band. In band is the convential way, as a tone.

The problem in Only inbound calls answered by receptionist...
Internaly (Mailbox etc.) and outbound calls (to the bank or similar) all phones from any extensions can use dtmf and */#.


*EDIT* I just tried to make a call outbound to my self... The receptionist did actually accept the dial as requested then..
Is it so that its only GSM that does not can make the choise "dial 1 for... and 2 for... "?
 
Hi,

Can you check what Codec your VOIP provider is using? And check with them that the DTMF is actually being sent by them?

To do this, you could temporarily bypass 3CX, and route the inbound VOIP call to your VOIP phone. Then on the phone making the outbound call, press the DTMF buttons. Do you hear the tones clearly? If not DMTF phones are not being sent.
 
Just made an EDIT in my last post...

The log still shows

13:29:45.781 ??: from 'server:0/MediaServer':EP:00000033@ payload #13 doesn't correspond initial SDP datalen=1
13:29:45.546 ??: from 'server:0/MediaServer':DTMF detection off on EP:00000035@
13:29:45.452 ??: from 'server:0/MediaServer':DTMF detection off on EP:00000033@
13:29:45.390 ??: from 'server:0/MediaServer':EP:00000033@ payload #13 doesn't correspond initial SDP datalen=1

What codec my Provider uses i dont know. I can try to check up tomorrow (monday)

*EDIT*
Just found on providers Forum...

Codecs used in priority order...
1: G.711A-Law >> 100 kbit/s
2: G.711mu-Law >> 100 kbit/s
3: G.729a >> 8 kbit/s
4: G.729 >> 8 kbit/s
5: G.723.1 >> 5,3 kbit/s
 
Hi!

Also have the same problem in log:

??: from 'playweb:0/MediaServer':EP:0000000C@ payload #13 doesn't correspond initial SDP datalen=1

i'm using an italian provider ( SKYPHO ) , i'm using WinXp , and client SIP SIEMENS 450 IP.
However, it's working fine... but i don't understand that error!

any news?

tx a lot!
 
I think it has to do with the binding to the media server.

You can set this under the advanced options of the gateway and/or of the Extension

I disabled mine on the extension and enabled on the gateway/provider.

See how you go.
 
tx, for yr help!

it seems that nothing happens..
Also try all combination , but i have also the error!

tx, however!
 
??: from 'playweb:0/MediaServer':EP:0000000C@ payload #13 doesn't correspond initial SDP datalen=1

I do no think that is provider related. There is something going on with the 3cx config.

Not sure what it is, but focus on the IPbpx and the phones.

What phones are you using?
What type of calls are your trying to make? Eg extension to extension, outbound, inbound etc.
 
itfarmer said:
??: from 'playweb:0/MediaServer':EP:0000000C@ payload #13 doesn't correspond initial SDP datalen=1

I do no think that is provider related. There is something going on with the 3cx config.

Not sure what it is, but focus on the IPbpx and the phones.

What phones are you using?
What type of calls are your trying to make? Eg extension to extension, outbound, inbound etc.

The error comes out only on outbound/inbound calls, not with extension -> extension.

im'trying with a SIEMENS C450 IP phone, and 3cx phone... only one extension, but the result is the same!

tx for yr help
 
Hi,

You can just ignore that message. It happens because a device detects silence and sends 'Comfort noice' payload (#13) instead. We're not supporting CN payload, but it doesn't affect normal audio flow anyway.

Best regards,
 
thanks a lot,

In fact calls are perfect!
now i can be quite :)

bye!
 
So.. Back to me then... :wink:

admin said:
Hi,

Can you check what Codec your VOIP provider is using? And check with them that the DTMF is actually being sent by them?

To do this, you could temporarily bypass 3CX, and route the inbound VOIP call to your VOIP phone. Then on the phone making the outbound call, press the DTMF buttons. Do you hear the tones clearly? If not DMTF phones are not being sent.

The answer to that is specified a bit further up in threas..

Anyone have a clue why my receptionist relays on some calls, but not everyone?
 
Technically,
digital receptionist can receive DTMF signals using the following mechanisms:
1. SIP INFO - if remote party sends it
2. RTP(RFC 2833) - if remote party sends it
3. Recognize DTMF tones in G.711A-Law or G.711mu-Law streams - if remote party sends it.

if remote phone sends in-band DTMF signal, it must work (if G.711A-Law or G.711mu-Law codec is used) because DTMF tones is a part of audio stream. VoIP provider will not influence DTMF detection. Quality of detection depends on quality of audio stream.

if remote phone sends DTMF using RTP(RFC2833) or SIP INFO directly to 3CX PhoneSystem - it also must work (as well as local extension).

If remote phone sends SIP INFO using VoIP provider SIP proxy, it is up to VoIP provider to send these packet to our PBX or ignore such kinds of packets.

If VoIP provider uses RTP proxy, RTP packets (RFC2833) will also be under VoIP provider control.

In case of local calls - there is no intermediate proxies between extension and 3CX Phone system and it works.

To investigate exact reasons, we need "Verbose" type of phone system logs.

Thank You for infomation.
Stepan
 
Thnx SY for mail-response.

Im still is a bit confused, and after installing the update 1928 i havent been able to do so much because the trouble to gett 3cx and IIS to coexist.
http://www.3cx.com/forums/viewtopic.php?t=1062

I have tried to get i touch with my IP provider for support and answers on how they relay on the DTMF issue, both mail and forum, but i dont get any response form there. :evil:
Actually considering to change provider now.

However, i still have the same trouble with receptionist dosent recognise incomming DTMF.
 
Hi Mori,

You said that problem with DTMFs is not common. Sometimes it works, but sometimes doesn't. Could you please describe scenario, when DTMF is not working, more precisely and send me your description? You know my email.

Thank you very much for your help

Stepan
 
There is no other way putting this than some times it works and sometimes it dont. :D

No really... None cellular phones (that i know obout) has managed to get response from receptionist when dialing to make there choice.

About PSTN and VoIP phones its almost like the lottery who works and who doesnt..
(Some callers CAN make a choice and the receptionist understands it, but with the other callers dont get a response, like cellular phones)

Dont think i can explain it more specific.. 8)
 
roger. :)

Thank you for information

Stepan
 
DTMF recognition specifics with 3CX Phone System:

3CX Phone System Digital Receptionist and VoiceMail systems will recognise incoming DTMF under the following scenarios:

1. Incoming stream delivers DTMF signals in-audio using G711a or G711u codecs - in this case the 3CX Phone System Media Server listens to the audio stream, and will detect DTMF signals and pass them through to DR or VM. Efficiency of DTMF detection depends on audio quality. Dropped packets will also reduce audio quality.
2. Incoming stream delivers DTMF signals out-of-audio using either SIP-INFO or RFC-2833 mechanism, using G711a or G711u or GSM codecs - in this case the DTMF signals are sent seperately from the actual audio stream, and are passed through to the DR or VM.

Note also that:

3. GSM is generally considered incapable of delivering DTMF signals, and detection mechanisms more often than not incorrectly detect signals. This is related to the way the codec algorithm compresses the audio stream, and is typically not system dependent. Due to these considerations, the 3CX Phone System will NOT attempt to detect DTMF signals in-audio when GSM is the audio codec in use.
4. Codecs other than G711a or G711u or GSM are not available for communication with the 3CX Phone System. Incoming calls originating from devices or providers attempting to negotiate other codecs, without providing one of the above 3 as a fallback, will be unable to deliver calls to the DR or VM - typically resulting in a dropped call.

Hope that this helps to make things clearer...
 
Mori,

Not sure what type of kit you are running. But it might be that some of your customers do not press the button long enough.

Try to set the DTMF sample rate to 3520.
 
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