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We have 25 D385 Deskphones and 17 M70 Dect with M900.

We also have several doors they are connected trough phonelines to our PBX. To open the door we have to send "#988" while in a call from the phone for example. The M70 sends DTMF tones, the D385 doesn't.

I tried the Settings "DTMF via SIP-INFO" to ON but without success. Also i switched RTP-Audio-Encryption to OFF, "Long SDP Answer" to ON and choose only these codecs "g722,pcma,pcmu,telephone-event". Nothing helped.


We have a IVR on our PBX, so i can test both phones. From Dect i can choose an option, from D385 not.

D385 Firmware is 10.1.169.13

What i'm doing wrong?


Regards.


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Thomas Schulze

End user

Joined: 17.05.2023

4 Comments

  1. Snom Gianmaria Tononi

    Hi,

    DTMF can only be changed by using the parameter user_dtmf_info, which you already configured to "on": depending on what the PBX expects, they need to be set accordingly on the phones too.

    To investigate this deeper, looking into what the phone sends and what the PBX replies, we need a private ticket: you can raise one if you are a Snom partner or you can ask your Snom reseller to do so for you.

    To find a partner that could help please contact our sales team Sales

    Thank you

  2. End user Thomas Schulze

    According to this site both partner negotiate what kind of DTMF to use on an invite. So, my freePBX (Asterisk) sends this to my D385 on a call:


    Message Body
            Session Description Protocol
                Session Description Protocol Version (v): 0
                Owner/Creator, Session Id (o): - 1739526995 1739526995 IN IP4 xx.xx.xx.xx
                Session Name (s): Asterisk
                Connection Information (c): IN IP4 xx.xx.xx.xx
                Time Description, active time (t): 0 0
                Media Description, name and address (m): audio 30452 RTP/AVP 0 8 111 3 9 101
                Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute (a): rtpmap:111 G726-32/8000
                Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute (a): rtpmap:9 G722/8000
                Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute (a): fmtp:101 0-16
                Media Attribute (a): ptime:20
                Media Attribute (a): maxptime:150
                Media Attribute (a): sendrecv
                [Generated Call-ID: 6798620e-892e-4f27-9071-1905d6a51ff1]


    As you can see, fmtp:101 0-16 is accepted, means out of band RFC 2833. And according to the description from SNOM's site, if it's acceptable for the Partner (in this case D385) it should answer in a SIP 200 OK with a=fmtp:101 0-11. But the answer from the D385 in a OK looks like this:


    Message Body
            Session Description Protocol
                Session Description Protocol Version (v): 0
                Owner/Creator, Session Id (o): root 234305972 234305973 IN IP4 xx.xx.xx.xx
                Session Name (s): call
                Connection Information (c): IN IP4 xx.xx.xx.xx
                Time Description, active time (t): 0 0
                Media Description, name and address (m): audio 56412 RTP/AVP 0 8 9
                Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute (a): rtpmap:9 G722/8000
                Media Attribute (a): ptime:20
                Media Attribute (a): sendrecv
                [Generated Call-ID: 6798620e-892e-4f27-9071-1905d6a51ff1]


    I don't see a Media Attribute fmtp, which means to me D385 don't accept DTMF RFC 2833, right?


    Regards

  3. End user Thomas Schulze

    Just to compare it to our M70 with M900. This is the Invite from our PBX:

        Message Body
            Session Description Protocol
                Session Description Protocol Version (v): 0
                Owner/Creator, Session Id (o): - 444818296 444818296 IN IP4 xx.xx.xx.xx
                Session Name (s): Asterisk
                Connection Information (c): IN IP4 xx.xx.xx.xx
                Time Description, active time (t): 0 0
                Media Description, name and address (m): audio 22224 RTP/AVP 0 8 111 3 9 101
                Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute (a): rtpmap:111 G726-32/8000
                Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute (a): rtpmap:9 G722/8000
                Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute (a): fmtp:101 0-16
                Media Attribute (a): ptime:20
                Media Attribute (a): maxptime:150
                Media Attribute (a): sendrecv
                [Generated Call-ID: 0687fd0f-b17d-4dc1-90a5-919c6c845452]



    And this is the 200 OK from the Dect (M70):

        Message Body
            Session Description Protocol
                Session Description Protocol Version (v): 0
                Owner/Creator, Session Id (o): 63 1258154656 1258154656 IN IP4 xx.xx.xx.xx
                Session Name (s): -
                Connection Information (c): IN IP4 xx.xx.xx.xx
                Time Description, active time (t): 0 0
                Media Description, name and address (m): audio 50018 RTP/AVP 0 101
                Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute (a): fmtp:101 0-15
                Media Attribute (a): sendrecv
                Media Attribute (a): rtcp:50019
                [Generated Call-ID: 0687fd0f-b17d-4dc1-90a5-919c6c845452]



  4. End user Thomas Schulze

    Problem solved. I don't understand it really, and it's a bit weird.

    I use an other LAN (which only have a different IP-Range) and it's working.