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Showing results for tags 'DTMF'.
I have found several threads about having a pause added so that a call can be made and then a DTMF tone played when answered. I was wondering is this has been accomplished. I have a new customer that had this ability on there old phone system. I have been able to get this to work with the Snom D725 by manually configuring a button for this, but would like to be able to have this in the PBX address book.
Hello, I have tried to use HD telephony codec (G722), quality is better however it seems that DTMF are not working. Phone : SPA525G PBX Vodia latest version For instance, when I try to call my voicemail, i enter my PIN but nothing happens. Thanks for help, Alexandre
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:  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  2013/09/10 13:19:40: Received DTMF *, call type attendant  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:email@example.com: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:firstname.lastname@example.org: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:email@example.com: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:firstname.lastname@example.org:5067> Remote-Party-ID: <sip:email@example.com: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 - - - -