PeterInFrankfurt Posted September 10, 2013 Report Share Posted September 10, 2013 I need to send Out-of-Band DTMF to an extension, but the snomone appears to transcode the DTMF to in-band. (The "extension" is really an IKON gateway which only supports out-of-band DTMF.) Here's a part of the logfile: [6] 2013/09/10 13:19:40: Call port 121: Different Codecs (local telephone-event/8000, remote PCMA/8000), callid f419f99b@pbx, falling back to transcoding [6] 2013/09/10 13:19:40: Received DTMF *, call type attendant [8] 2013/09/10 13:19:40: Attendant: Ignoring the DTMF * in the state connected How can I make the pbx forward the DTMF as an RTP-event to the "extension"? In case it's relevant, here's the INVITE etc. 2013/9/10 13:19:39 Tx: udp:172.27.66.105:5067 (959 bytes) INVITE sip:240@172.27.66.105:5067 SIP/2.0 Via: SIP/2.0/UDP 172.27.66.106:5060;branch=z9hG4bK-abb91ea6476deec2bab243dd796c1b78;rport From: "Svalbard MER1" <sip:401@localhost;user=phone>;tag=51546 To: "MSG Control Room" <sip:240@localhost> Call-ID: 57a2ef9c@pbx CSeq: 27025 INVITE Max-Forwards: 70 Contact: <sip:240@172.27.66.106:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.1.0 Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 329 v=0 o=- 63349 63349 IN IP4 172.27.66.106 s=- c=IN IP4 172.27.66.106 t=0 0 m=audio 41036 RTP/AVP 8 0 9 2 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv and here's the OK back 2013/9/10 13:19:39 Rx: udp:172.27.66.105:5067 (820 bytes) SIP/2.0 200 OK Via: SIP/2.0/UDP 172.27.66.106:5060;branch=z9hG4bK-abb91ea6476deec2bab243dd796c1b78;rport From: "Svalbard MER1" <sip:401@localhost;user=phone>;tag=51546 To: "MSG Control Room" <sip:240@localhost>;tag=802229 Call-ID: 57a2ef9c@pbx CSeq: 27025 INVITE Contact: <sip:240@172.27.66.105:5067> Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE Content-Length: 149 Content-Type: application/sdp Server: (innovaphone H323/[unknown]) Supported: replaces,100rel,sec-agree,answermode P-Preferred-Identity: <sip:240@172.27.66.105:5067> Remote-Party-ID: <sip:240@172.27.66.105:5067>;party=called;screen=no;privacy=off v=0 o=- 49 1 IN IP4 172.27.66.105 s=- c=IN IP4 172.27.66.105 t=0 0 a=sendrecv m=audio 8184 RTP/AVP 8 0 a=ptime:20 a=silenceSupp:off - - - - Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 10, 2013 Report Share Posted September 10, 2013 Well the SDP in the response says "okay, you can send me aLaw and uLaw"; but nothing about telephone-event (which is RFC2833 a.k.a. RFC4733 out of band). Maybe there is a setting on the gateway that needs to be turned on to offer the DTMF codec as well? Quote Link to comment Share on other sites More sharing options...
PeterInFrankfurt Posted September 11, 2013 Author Report Share Posted September 11, 2013 Thanks for the reply. I can find no relevant settings on the gateway. I noticed that a=fmtp:101 0-16 is not mentioned in the gateway's reply. However, RFC 2833 says: Since all implementations MUST be able to receive events 0 through 15, listing these events in the a=fmtp line is OPTIONAL. Are you saying the snomone pbx insists the gateway response explicitly includes a=fmtp... ? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 11, 2013 Report Share Posted September 11, 2013 snom ONE does not need the FMTP line, but it needs a a-line like a=rtpmap:101 telephone-event/8000. The FMTP line provides additional information about a codec (parameters). The one above looks like a telephone-event parameter (DTMF), and it says send everything from 0-9, #, *, A, B, C, D and also hook-flash (according to RFC2833). But anyway this is the right corner. If it is sending the fmtp line, it actually must also send the a=rtpmap:101 line. Quote Link to comment Share on other sites More sharing options...
PeterInFrankfurt Posted September 11, 2013 Author Report Share Posted September 11, 2013 I've found a way to make the gateway behave correctly - it works now! Many thanks for your help. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 11, 2013 Report Share Posted September 11, 2013 Can you tell us what the trick was? Then others may find this here on the forum and don't have to go through a similar experience... Quote Link to comment Share on other sites More sharing options...
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