[OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)

2014-01-29 Thread Vignesh Sethuraman
Hello All, 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

Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)

2014-01-29 Thread Somphol Boonjing
On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.comwrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP

Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)

2014-01-29 Thread Justin Carney
I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered

Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)

2014-01-29 Thread moataz_mmdh
Hello What do you see when you do 'debug ccsip messages' on cucme Sent using BlackBerry® from mobinil -Original Message- From: Vignesh Sethuraman sethuvign...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com Date: Wed, 29 Jan 2014 22:48:46 To: