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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Jamie Rees<br><b>Sent:</b> Monday, July 06, 2015 5:54 PM<br><b>To:</b> asterisk-users@lists.digium.com<br><b>Subject:</b> [asterisk-users] DTMF issue<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-GB>Hello folks,<br><br>We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. <br><br>I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility, but this hasn't helped at all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not clearly state which setting this is on the handsets.<br><br>I have enabled DTMF logging and spoken to the SIP provider, but they couldn't really help much. I presume the issue is local to our phone system but other than the logs below, have nothing to go on:<br><br>[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF begin '2' received on SIP/sip-out-00021c6d<br>[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF begin passthrough '2' on SIP/sip-out-00021c6d<br>[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF end '2' received on SIP/sip-out-00021c6d, duration 200 ms<br>[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF end accepted with begin '2' on SIP/sip-out-00021c6d<br>[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF end passthrough '2' on SIP/sip-out-00021c6d<br>[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF begin '3' received on SIP/209-00021cac<br>[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF begin passthrough '3' on SIP/209-00021cac<br>[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end '3' received on SIP/209-00021cac, duration 90 ms<br>[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end accepted with begin '3' on SIP/209-00021cac<br>[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end '3' detected to have actual duration 78 on the wire, emulation will be triggered on SIP/209-00021cac<br>[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end '3' has duration 78 but want minimum 80, emulating on SIP/209-00021cac<br>[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end emulation of '3' queued on SIP/209-00021cac<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Can someone please provide any tips? <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Thanks,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>Jamie <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB style='color:#1F497D'>This doesn’t help, but It DOES sound familiar. I’ve not seen this for a long time. If I can remember I’ll write back. Just thought I’d let you know you’re not crazy. </span><span lang=EN-GB style='font-family:Wingdings;color:#1F497D'>J</span><span lang=EN-GB style='color:#1F497D'><o:p></o:p></span></p></div></body></html>