<p dir="ltr">Let me try with dtmfmode as auto... </p>
<div class="gmail_quote">On 28 May 2013 19:32, "Asghar Mohammad" <<a href="mailto:asghar144@gmail.com">asghar144@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">work around was block dtmf.<div>set wrong type of dtmf in incoming trunk.</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N <span dir="ltr"><<a href="mailto:gopalakrishnan.an@gmail.com" target="_blank">gopalakrishnan.an@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">So any resolution for this?<div><br></div><div>I suspect it could be related to RE INVITE</div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad <span dir="ltr"><<a href="mailto:asghar144@gmail.com" target="_blank">asghar144@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">i had this in past there was an ATA configured to send 9 at the end of dialing in my case.</div><div class="gmail_extra">
<br><br><div class="gmail_quote"><div><div>On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <span dir="ltr"><<a href="mailto:gopalakrishnan.an@gmail.com" target="_blank">gopalakrishnan.an@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Hi,<div><br></div><div>I am receiving DTMF without any reason after call establishment.</div>
<div><br></div>
<div>The log as follows, and I suspect something related to directmedia,</div><div>
<div>[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48</div><div>[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48</div>
<div>[May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on SIP/MyTrunk-000a4b49, duration 0 ms</div><div>[May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin '*' on SIP/MyTrunk-000a4b49</div>
<div>[May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on SIP/MyTrunk-000a4b49</div><div>[May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on SIP/MyTrunk-000a4b49, duration 0 ms</div>
<div>[May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin '8' on SIP/MyTrunk-000a4b49</div><div>[May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on SIP/MyTrunk-000a4b49</div>
<div>[May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on SIP/MAN-000a4af0, duration 100 ms</div><div>[May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with duration 100 queued on SIP/MAN-000a4af0</div>
<div>[May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued on SIP/MAN-000a4af0</div><div>[May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on SIP/MAN-000a4b41, duration 100 ms</div>
<div>[May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with duration 100 queued on SIP/MAN-000a4b41</div><div>[May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued on SIP/MAN-000a4b41</div>
<div>[May 17 00:33:55] VERBOSE[4106] pbx.c: == Spawn extension (sip-trunk-inbound, <a href="tel:2127773456" value="+12127773456" target="_blank">2127773456</a>, 1) exited non-zero on 'SIP/MyTrunk-000a4af3'</div>
<div>[May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1] NoOp("SIP/MAN-000a4b09", "16") in new stack</div>
<div>[May 17 00:33:56] VERBOSE[4136] pbx.c: == Spawn extension (trunk-outbound, 777787457712, 2) exited non-zero on 'SIP/MAN-000a4b09'</div><div><br></div><div>Is this some thing related to SIP RE-INVITE?</div>
<div><br></div><div>Thanks. </div><div><br></div></div></div><span><font color="#888888">
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