<html><body><div style="color:#000; background-color:#fff; font-family:arial, helvetica, sans-serif;font-size:10pt">sure Brian. Please find attached, the debug output.<br><br>At the end you will find MWI notification message, those are received only when I hang up the phone and not when I press # key to send the message. Just for your clarification.<br><div> </div><div>Thanks & Regards,<br>Vignesh S</div><div style="display: block;" class="yahoo_quoted"> <br> <br> <div style="font-family: arial, helvetica, sans-serif; font-size: 10pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div dir="ltr"> <font face="Arial" size="2"> On Wednesday, 29 January 2014 11:25 PM, Brian Meade (brmeade) <brmeade@cisco.com> wrote:<br> </font> </div> <div class="y_msg_container"><div id="yiv7526524843"><style>#yiv7526524843 #yiv7526524843 --
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<div class="yiv7526524843MsoNormal"><span style="font-size:11.0pt;">Can you collect a “debug ccsip messages” for one of the calls?</span></div>
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<div class="yiv7526524843MsoNormal"><b><span style="font-size:10.0pt;">From:</span></b><span style="font-size:10.0pt;"> vignesh sethuraman [mailto:sethuvignesh@yahoo.co.in]
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<b>Sent:</b> Wednesday, January 29, 2014 5:24 PM<br clear="none">
<b>To:</b> Brian Meade (brmeade); cisco-voip@puck.nether.net<br clear="none">
<b>Subject:</b> Re: [cisco-voip] DTMF Issue with SIP TRUNK (CUCM -- SIP Trunk---- CUCME<---CUE)</span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;">Hello Brian,<br clear="none">
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I tried that now and reset the SIP trunk but no luck.<br clear="none">
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My CUCM version is 8.6.2. and CUCME is running 15.2(1)T3 and my CUE is running 8.6.7.</span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;">Thanks & Regards,<br clear="none">
Vignesh S</span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;">On Wednesday, 29 January 2014 11:17 PM, Brian Meade (brmeade) <<a rel="nofollow" shape="rect" ymailto="mailto:brmeade@cisco.com" target="_blank" href="mailto:brmeade@cisco.com">brmeade@cisco.com</a>> wrote:</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:11.0pt;color:black;">Viki,</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:11.0pt;color:black;">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”.</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:11.0pt;color:black;">Brian</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><b><span style="font-size:10.0pt;color:black;">From:</span></b><span style="font-size:10.0pt;color:black;"> cisco-voip [<a rel="nofollow" shape="rect" ymailto="mailto:cisco-voip-bounces@puck.nether.net" target="_blank" href="mailto:cisco-voip-bounces@puck.nether.net">mailto:cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>vignesh sethuraman<br clear="none">
<b>Sent:</b> Wednesday, January 29, 2014 5:07 PM<br clear="none">
<b>To:</b> <a rel="nofollow" shape="rect" ymailto="mailto:cisco-voip@puck.nether.net" target="_blank" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br clear="none">
<b>Subject:</b> [cisco-voip] DTMF Issue with SIP TRUNK (CUCM -- SIP Trunk---- CUCME<---CUE)</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;color:black;">Hello All,</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;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.</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;color:black;">Here is my scenario and the configuration.</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;color:black;">(PhoneA) -- CUCM ----SIP TRUNK ----CUCME (PhoneD) ---> CUE.<br clear="none">
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voice service voip<br clear="none">
ip address trusted list<br clear="none">
ipv4 0.0.0.0 0.0.0.0<br clear="none">
allow-connections h323 to h323<br clear="none">
allow-connections h323 to sip<br clear="none">
allow-connections sip to h323<br clear="none">
allow-connections sip to sip<br clear="none">
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none<br clear="none">
h323<br clear="none">
sip<br clear="none">
bind control source-interface Loopback0<br clear="none">
bind media source-interface Loopback0<br clear="none">
registrar server expires max 600 min 60<br clear="none">
!<br clear="none">
voice class codec 1<br clear="none">
codec preference 1 g711ulaw<br clear="none">
codec preference 2 g729r8<br clear="none">
!<br clear="none">
telephony-service<br clear="none">
sdspfarm units 3<br clear="none">
sdspfarm transcode sessions 4<br clear="none">
max-ephones 5<br clear="none">
max-dn 10<br clear="none">
ip source-address 3.3.3.3 port 2000<br clear="none">
load 7945 term45.default.loads<br clear="none">
time-zone 28<br clear="none">
time-format 24<br clear="none">
date-format dd-mm-yy<br clear="none">
voicemail 3600<br clear="none">
max-conferences 8 gain -6<br clear="none">
call-forward pattern .T<br clear="none">
moh "music-on-hold.au"<br clear="none">
dn-webedit <br clear="none">
transfer-system full-consult<br clear="none">
transfer-pattern .T<br clear="none">
create cnf-files <br clear="none">
!<br clear="none">
sccp local Loopback0<br clear="none">
sccp ccm 10.0.10.160 identifier 2 version 7.0 <br clear="none">
sccp ccm 3.3.3.3 identifier 1 version 7.0 <br clear="none">
sccp ip precedence 3<br clear="none">
sccp<br clear="none">
sccp ccm group 1<br clear="none">
associate ccm 1 priority 1<br clear="none">
associate ccm 2 priority 2<br clear="none">
associate profile 1 register BR2-IOS-XCODE<br clear="none">
associate profile 2 register BR2-IOS-CFB<br clear="none">
keepalive retries 5<br clear="none">
switchover method immediate<br clear="none">
switchback method immediate<br clear="none">
switchback interval 15<br clear="none">
!<br clear="none">
dspfarm profile 1 transcode <br clear="none">
codec g729r8<br clear="none">
codec g711ulaw<br clear="none">
codec g711alaw<br clear="none">
codec g729ar8<br clear="none">
codec g729abr8<br clear="none">
maximum sessions 4<br clear="none">
associate application SCCP<br clear="none">
!<br clear="none">
dspfarm profile 2 conference <br clear="none">
codec g711ulaw<br clear="none">
codec g711alaw<br clear="none">
codec g729ar8<br clear="none">
codec g729abr8<br clear="none">
codec g729r8<br clear="none">
codec g729br8<br clear="none">
maximum sessions 2<br clear="none">
associate application SCCP<br clear="none">
!<br clear="none">
dial-peer voice 1000 voip<br clear="none">
destination-pattern [15]...$<br clear="none">
session protocol sipv2<br clear="none">
session target ipv4:10.0.10.160<br clear="none">
incoming called-number .<br clear="none">
voice-class codec 1 <br clear="none">
dtmf-relay sip-notify<br clear="none">
no vad<br clear="none">
!<br clear="none">
dial-peer voice 3600 voip<br clear="none">
destination-pattern 3[16]00$<br clear="none">
session protocol sipv2<br clear="none">
session target ipv4:10.10.202.100<br clear="none">
incoming called-number 399[89]....<br clear="none">
dtmf-relay sip-notify<br clear="none">
codec g711ulaw<br clear="none">
no vad<br clear="none">
!<br clear="none">
<b>On the CUCM, I did the following,<br clear="none">
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</b>Media Termination Point Required (Checked)<br clear="none">
MTP Preferred Originating CodecRequired Field: g711ulaw<br clear="none">
DTMF Signaling MethodRequired Field: No preference<br clear="none">
Non Secure SIP Trunk Profile: I am using TCP+UDP for INCOMING + Accept Unsolicited Notification (Checked).</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;color:black;">Please let me know what I am missing.</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;color:black;">Thanks,</span><span style="color:black;"></span></div>
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<div class="yiv7526524843MsoNormal" style="background:white;"><span style="font-size:10.0pt;color:black;">Viki</span><span style="color:black;"></span></div>
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