For the first issue, if you add the CME router as an H323 gateway in CUCM the correct bandwidth will show. Make sure that the CSS includes the partition that contains the phones.
From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David A Sent: Monday, October 04, 2010 1:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls Hi All, I am doing the Vol2 Lab2 GK scenario and running into a couple of issues. issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I check the codec used on the call on both phones it says g729. The gk-tunk is in DP GK with region g729 to everyone. 2811-HQ-GW#sh gatekeeper call Total number of active calls = 1. GATEKEEPER CALL INFO ==================== LocalCallID Age(secs) BW 25-49659 21 128(Kbps) <------- should be 16kbps as per the requirement Endpt(s): Alias E.164Addr src EP: SiteC-GW 3003 CallSignalAddr Port RASSignalAddr Port 10.10.110.3 1720 10.10.110.3 58555 Endpt(s): Alias E.164Addr dst EP: gk-trunk_2 1#1002 CallSignalAddr Port RASSignalAddr Port 10.137.151.26 1720 10.137.151.26 33447 issue 2 - Call from CUCM to CME SIP phone(g711) works fine and invokes the transcoder and i see a 16kbps GK call. However when I call from CME SIP phone to any CUCM phone, CUCM phone rings and I can answer it. However it drops after a few seconds and I see no transcoder being used. Here are my configs Site C - voice register pool 1 id mac 0025.4593.0368 type 7975 number 1 dn 1 number 2 dn 2 template 1 description 32143002 codec g711ulaw ! dial-peer voice 15 voip destination-pattern [15]...$ session target ras incoming called-number . tech-prefix 1# dtmf-relay h245-alphanumeric rtp-nte ! dial-peer voice 3000 voip incoming called-number 3...$ dtmf-relay h245-alphanumeric ! Any clues? Thanks, DA
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