Hi,

we are experiencing problems with DTMF between Snom Sip Phones and FS. The DMTF Type is set to SIP INFO (sip info only on Snom Phones). We could nail it down to a missing "telephone-event" parameter in sip sdp:
INVITE sip:***6@spklw.x;user:phone SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-15eyqwai33r1;rport
From: "Test" <sip:8000@spklw.x>;tag=qqnbrunjmq
To: <sip:***6@spklw.x;user:phone>
Call-ID: 3c33e6c729b3-t9yol2613q6v
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:8000@ip:5060;line=9lwqdnrx>;reg-id=1
X-Serialnumber: 000413262C33
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/8.4.32
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 334284706 334284706 IN IP4 phoneip
s=call
c=IN IP4 phoneip
t=0 0
m=audio 12610 RTP/AVP 8 9
a=direction:both
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a:ptime:20
a=rtcp-xr:stat-summary=loss,dup,jitt
a=sendrecv


That resulted in sofia_glue.c disabling DTMF. A slight change in sofia_glue.c (Line 4839 in commit 71964f61ac4817ecfc516edaf4b643f41236868d) did the trick for us:
if (tech_pvt->dtmf_type == DTMF_INFO) { //EWETEL
} else {                                //EWETEL
                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Disable 2833 dtmf\n");
                switch_channel_set_variable(tech_pvt->channel, "dtmf_type", "none");
                tech_pvt->dtmf_type = DTMF_NONE;
                te = tech_pvt->recv_te = 0;
} //EWETEL


Now the question is:
Is this a too restrictive behaviour in FS or a shortcoming from the Snom phones (which would mean we have to submit a bug, wait, wait, wait and then hopefully someday it gets fixed :) )?

Regards,
Mitja

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Mitja Thomas
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