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<body class='hmmessage'><div dir='ltr'>Hi Anthony,<BR> <BR>I have installed I took CUCM trace and used Translator X, when I scroll down to the bottom, under DTMF Method Originator details i see: (3) Out of Band and RFC2833.<BR>Under Destination details i see:(1) Out of Band (KPML / NOTIFY)<BR> <BR>This is on a call that did not work when being asked to press 1 for etc..<BR> <BR>Regards<BR> <BR>Costas<br> <BR><div><hr id="stopSpelling">Date: Wed, 14 May 2014 10:38:24 -0500<br>Subject: Re: [cisco-voip] DTMF tone<br>From: avholloway+cisco-voip@gmail.com<br>To: ckos1976@hotmail.com<br>CC: cisco-voip@puck.nether.net<br><br><div dir="ltr">First and foremost, you cannot have DTMF relay on a POTS dial peer. DTMF relay is for IP networks. On the POTS side, the DSP will turn the DTMF relay from CUCM (H.245 alphanumeric) or the endpoint (RTP-NTE) into actual audio.<div>
<div><br></div><div>Source: <a href="http://www.cisco.com/c/en/us/td/docs/ios/voice/command/reference/vr_book/vr_d2.html#wp1449003" target="_blank">http://www.cisco.com/c/en/us/td/docs/ios/voice/command/reference/vr_book/vr_d2.html#wp1449003</a></div>
<div><br></div><div>The command voice rtp send-recv command only helps for calls to the PSTN where the IVR is in the cloud and occurs before the ISDN connect message. This is also referred to as early media. Is that what's happening on your bad IVR calls?</div>
<div><br></div><div>Source: <a href="http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_H.323_Gateway_Troubleshooting#No_DTMF_Digits_or_Audio_Passed_on_VoIP_Calls_to_PSTN_or_PBX" target="_blank">http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_H.323_Gateway_Troubleshooting#No_DTMF_Digits_or_Audio_Passed_on_VoIP_Calls_to_PSTN_or_PBX</a></div>
<div><br></div><div>I see that your incoming dial peer from CUCM (incoming called-number .T) has a voice-class h323 1 on it, but you did not include the details for it. Also, you don't need .T you only need . for the incoming called-number command. Could you show us your h323 class config?</div>
<div><br></div><div>Source: <a href="http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_Call_Flow_Overview#Inbound_Dial_Peer_Matching" target="_blank">http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_Call_Flow_Overview#Inbound_Dial_Peer_Matching</a></div>
<div><br></div><div>Also, consider pulling CallManager traces to see which DTMF relay was negotiated on both good and bad calls, since you have both out of band and in band specified. This will help to determine where the problem could be, because out of band DTMF relay takes a different path and has different device dependencies than does in band. If you haven't used TranslatorX yet, it's awesome and you should. Drag and drop your trace files in to it, click the Call List button, filter and find your bad call, double click it, and then scroll to the bottom of the summary page and it will tell you the DTMF negotiation.</div>
<div><br></div><div>Source: <a href="http://translatorx.cisco.com/" target="_blank">http://translatorx.cisco.com/</a></div><div><br></div><div>Since you do have both out of band and in band, I would suspect MTP's are being inserted to correct DTMF relay, but your traces would confirm if good calls are using MTPs and bad calls are not. CUCM will save a call which failed to allocate an MTP by default, but you run the risk of diminished supplementary services such as DTMF.</div>
<div><br></div><div>This is from the CUCM 8x SRND (you didn't say which version of CUCM you have):</div><div><br></div><div>FYI The reference to NTE is the same as rtp-nte</div></div><blockquote style="padding: 0px; border: currentColor;">
<div><div><h3 style="color: rgb(0, 0, 0); font-family: Arial,Helvetica,sans-serif; font-size: 13px;">DTMF Relay on H.323 Gateways and Cisco Unified Border Element</h3></div></div><div><div><p style="color: rgb(0, 0, 0); font-family: Arial,Helvetica,sans-serif; font-size: 12px;">
H.323 gateways support DTMF relay via H.245 Alphanumeric, H.245 Signal, NTE, and audio in the media stream. The NTE option must not be used because it is not supported on Unified CM for H.323 gateways at this time. The preferred option is H.245 Signal. MTPs are required for establishing calls to an H.323 gateway if the other endpoint does not have signaling capability in common with Unified CM. For example, a Cisco Unified IP Phone 7960 running the SIP stack supports only NTEs, so an MTP is needed with an H.323 gateway.</p>
</div></div></blockquote><div><div><a style="color: rgb(0, 0, 0); font-family: Arial,Helvetica,sans-serif; font-size: 13px;" name="wp1233307" target="_blank"></a><span style="color: rgb(0, 0, 0); font-family: Arial,Helvetica,sans-serif; font-size: 13px;"></span></div>
<div><br></div><div>Source: <a href="http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/media.html#wpmkr1130352" target="_blank">http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/media.html#wpmkr1130352</a></div>
<div><br></div><div>Summary:</div><div><ol><li>No DTMF relay on POTS dial peers</li><li>incoming called-number .</li><li>show run | section h323 1</li><li>CCM Traces to confirm DTMF relay negoatiation</li><li>CCM Traces to confirm MTP (or XCODER) allocation</li>
<li>Should use H.245 signaling and not H245 alphanumeric and no rtp-nte according to SRND</li></ol><div>I hope that was helpful. Sorry there wasn't an easy answer to your query, but DTMF relay, if not designed for, can be a roll of the dice.</div>
</div></div></div><div class="ecxgmail_extra"><br><br><div class="ecxgmail_quote">On Wed, May 14, 2014 at 4:26 AM, costas georgiou <span dir="ltr"><<a href="mailto:ckos1976@hotmail.com" target="_blank">ckos1976@hotmail.com</a>></span> wrote:<br>
<blockquote style="padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="ecxgmail_quote">
<div><div dir="ltr">Hi all,<br> <br>I have a customer who says a handfull of users are having difficulties accessing an IVR externally. When they dial the number, they get the message "press 1 for this etc" but the DTMF tone does not always work so they use their mobiles. They are running on an H323 gateway. It already has the voice rtp send-recv command globally. Below is aa liat of the dial-peers. Will the DTMF command need to be on the outgoing Dial-peers?<br>
<br>dial-peer voice 1 pots<br> description *** Default inbound Dial-peer ***<br> incoming called-number .T<br> direct-inward-dial<br>!<br>dial-peer voice 100 pots<br> description *** Outbound PSTN Dial Peer ***<br> preference 1<br>
destination-pattern 9T<br> progress_ind setup enable 3<br> progress_ind alert enable 8<br> progress_ind progress enable 8<br> progress_ind disconnect enable 8<br> port 0/0/0:15<br>!<br>dial-peer voice 200 voip<br> description *** Inbound VOIP Dial peer to CUCM ***<br>
preference 1<br> destination-pattern 272..<br> session target ipv4:x.x.x.x<br> voice-class codec 1<br> voice-class h323 1<br> dtmf-relay h245-alphanumeric rtp-nte<br> ip qos dscp cs3 signaling<br> no vad<br>!<br>dial-peer voice 201 voip<br>
description *** Inbound VOIP Dial peer to CUCM ***<br> preference 2<br> destination-pattern 272..<br> session target ipv4:x.x.x.x<br> voice-class codec 1<br> voice-class h323 1<br> dtmf-relay h245-alphanumeric rtp-nte<br>
ip qos dscp cs3 signaling<br> no vad<br>!<br>dial-peer voice 202 voip<br> description *** Inbound VOIP Dial peer to CUCM ***<br> preference 3<br> destination-pattern 272..<br> session target ipv4:x.x.x.x<br> voice-class codec 1<br>
voice-class h323 1<br> dtmf-relay h245-alphanumeric rtp-nte<br> ip qos dscp cs3 signaling<br> no vad<br>!<br>dial-peer voice 2 voip<br> description *** Inbound VOIP Dial peer FROM CUCM ***<br> incoming called-number .T<br>
voice-class codec 1<br> voice-class h323 1<br> dtmf-relay h245-alphanumeric rtp-nte<br> ip qos dscp cs3 signaling<br> no vad<br>!<br>Thanks <br> <br>Cos<br> </div></div>
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