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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Viki,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">What do you see getting negotiated for DTMF on the CUCM side? You might want to try forcing out-of-band on the CUCM side. Try changing the DTMF Signaling
Method to “OOB and RFC 2833”.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Brian<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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<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""> cisco-voip [mailto:cisco-voip-bounces@puck.nether.net]
<b>On Behalf Of </b>vignesh sethuraman<br>
<b>Sent:</b> Wednesday, January 29, 2014 5:07 PM<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> [cisco-voip] DTMF Issue with SIP TRUNK (CUCM -- SIP Trunk---- CUCME<---CUE)<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt;background:white"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Hello All,<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt;background:white"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">I am facing an issue with dtmf-relay. PhoneA registered to CUCM (SiteA) calls PhoneD registered to CUCME (Site C).
Between Regions G729 codec is negotiated. PhoneD call-forward no answer to Voicemail. CUE integrated with CME. After leaving the Voicemail from PhoneA to PhoneD, when I press # key to send the Voicemail, it is not recognized.<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt;background:white"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Here is my scenario and the configuration.<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt;background:white"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">(PhoneA) -- CUCM ----SIP TRUNK ----CUCME (PhoneD) ---> CUE.<br>
<br>
voice service voip<br>
ip address trusted list<br>
ipv4 0.0.0.0 0.0.0.0<br>
allow-connections h323 to h323<br>
allow-connections h323 to sip<br>
allow-connections sip to h323<br>
allow-connections sip to sip<br>
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none<br>
h323<br>
sip<br>
bind control source-interface Loopback0<br>
bind media source-interface Loopback0<br>
registrar server expires max 600 min 60<br>
!<br>
voice class codec 1<br>
codec preference 1 g711ulaw<br>
codec preference 2 g729r8<br>
!<br>
telephony-service<br>
sdspfarm units 3<br>
sdspfarm transcode sessions 4<br>
max-ephones 5<br>
max-dn 10<br>
ip source-address 3.3.3.3 port 2000<br>
load 7945 term45.default.loads<br>
time-zone 28<br>
time-format 24<br>
date-format dd-mm-yy<br>
voicemail 3600<br>
max-conferences 8 gain -6<br>
call-forward pattern .T<br>
moh "music-on-hold.au"<br>
dn-webedit <br>
transfer-system full-consult<br>
transfer-pattern .T<br>
create cnf-files <br>
!<br>
sccp local Loopback0<br>
sccp ccm 10.0.10.160 identifier 2 version 7.0 <br>
sccp ccm 3.3.3.3 identifier 1 version 7.0 <br>
sccp ip precedence 3<br>
sccp<br>
sccp ccm group 1<br>
associate ccm 1 priority 1<br>
associate ccm 2 priority 2<br>
associate profile 1 register BR2-IOS-XCODE<br>
associate profile 2 register BR2-IOS-CFB<br>
keepalive retries 5<br>
switchover method immediate<br>
switchback method immediate<br>
switchback interval 15<br>
!<br>
dspfarm profile 1 transcode <br>
codec g729r8<br>
codec g711ulaw<br>
codec g711alaw<br>
codec g729ar8<br>
codec g729abr8<br>
maximum sessions 4<br>
associate application SCCP<br>
!<br>
dspfarm profile 2 conference <br>
codec g711ulaw<br>
codec g711alaw<br>
codec g729ar8<br>
codec g729abr8<br>
codec g729r8<br>
codec g729br8<br>
maximum sessions 2<br>
associate application SCCP<br>
!<br>
dial-peer voice 1000 voip<br>
destination-pattern [15]...$<br>
session protocol sipv2<br>
session target ipv4:10.0.10.160<br>
incoming called-number .<br>
voice-class codec 1 <br>
dtmf-relay sip-notify<br>
no vad<br>
!<br>
dial-peer voice 3600 voip<br>
destination-pattern 3[16]00$<br>
session protocol sipv2<br>
session target ipv4:10.10.202.100<br>
incoming called-number 399[89]....<br>
dtmf-relay sip-notify<br>
codec g711ulaw<br>
no vad<br>
!<br>
<b>On the CUCM, I did the following,<br>
<br>
</b>Media Termination Point Required (Checked)<br>
MTP Preferred Originating CodecRequired Field: g711ulaw<br>
DTMF Signaling MethodRequired Field: No preference<br>
Non Secure SIP Trunk Profile: I am using TCP+UDP for INCOMING + Accept Unsolicited Notification (Checked).<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt;background:white"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Please let me know what I am missing.<o:p></o:p></span></p>
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<p class="MsoNormal" style="background:white"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Thanks,<o:p></o:p></span></p>
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<p class="MsoNormal" style="background:white"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Viki<o:p></o:p></span></p>
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